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.

 

Possible values of arg are:

-version

Show version.

-formats

Show available formats.

 

The fields preceding the format names have the following meanings:

-codecs

Show all codecs known to libavcodec.

 

Note that the term 'codec' is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format.

-decoders

Show available decoders.

-encoders

Show all available encoders.

-bsfs

Show available bitstream filters.

-protocols

Show available protocols.

-filters

Show available libavfilter filters.

-pix_fmts

Show available pixel formats.

-sample_fmts

Show available sample formats.

-loglevel loglevel | -v loglevel

Set the logging level used by the library. loglevel is a number or a string containing one of the following values:

AVOptions

These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the -help option. They are separated into two categories:

generic

These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.

private

These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.

For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the id3v2_version private option of the MP3 muxer:
avconv -i input.flac -id3v2_version 3 out.mp3
All codec AVOptions are obviously per-stream, so the chapter on stream specifiers applies to them Note -nooption syntax cannot be used for boolean AVOptions, use -option 0/-option 1. Note2 old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon.

Codec AVOptions

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

set bitrate (in bits/s)

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

Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to minimum/maximum bitrate. Lowering tolerance too much has an adverse effect on quality.

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

Possible values:

-sub_id[:stream_specifier] integer ()

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

set motion estimation method

 

Possible values:

-extradata_size[:stream_specifier] integer ()

-time_base[:stream_specifier] rational number ()

-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)

-frame_number[:stream_specifier] integer ()

-delay[:stream_specifier] integer ()

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

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

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

video quantizer scale blur (VBR)

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

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

-mv_bits[:stream_specifier] integer ()

-header_bits[:stream_specifier] integer ()

-i_tex_bits[:stream_specifier] integer ()

-p_tex_bits[:stream_specifier] integer ()

-i_count[:stream_specifier] integer ()

-p_count[:stream_specifier] integer ()

-skip_count[:stream_specifier] integer ()

-misc_bits[:stream_specifier] integer ()

-frame_bits[:stream_specifier] integer ()

-codec_tag[:stream_specifier] integer ()

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

work around not autodetected encoder bugs

 

Possible values:

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

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

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

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

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

how strictly to follow the standards

 

Possible values:

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

QP offset between P- and B-frames

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

set error detection flags

 

Possible values:

-has_b_frames[:stream_specifier] integer ()

-block_align[:stream_specifier] integer ()

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

use MPEG quantizers instead of H.263

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

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

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

experimental quantizer modulation

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

experimental quantizer modulation

-rc_override_count[:stream_specifier] integer ()

-rc_eq[:stream_specifier] string (output,video)

Set rate control equation. When computing the expression, besides the standard functions defined in the section 'Expression Evaluation', the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.

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

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

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

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

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

set ratecontrol buffer size (in bits)

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

currently useless

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

QP factor between P- and I-frames

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

QP offset between P- and I-frames

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

initial complexity for 1-pass encoding

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

DCT algorithm

 

Possible values:

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

compresses bright areas stronger than medium ones

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

temporal complexity masking

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

spatial complexity masking

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

inter masking

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

compresses dark areas stronger than medium ones

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

select IDCT implementation

 

Possible values:

-slice_count[:stream_specifier] integer ()

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

set error concealment strategy

 

Possible values:

-bits_per_coded_sample[:stream_specifier] integer ()

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

prediction method

 

Possible values:

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

sample aspect ratio

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

print specific debug info

 

Possible values:

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

visualize motion vectors (MVs)

 

Possible values:

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

full-pel ME compare function

 

Possible values:

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

sub-pel ME compare function

 

Possible values:

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

macroblock compare function

 

Possible values:

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

interlaced DCT compare function

 

Possible values:

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

diamond type & size for motion estimation

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

amount of motion predictors from the previous frame

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

pre motion estimation

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

pre motion estimation compare function

 

Possible values:

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

diamond type & size for motion estimation pre-pass

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

sub-pel motion estimation quality

-dtg_active_format[:stream_specifier] integer ()

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

limit motion vectors range (1023 for DivX player)

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

intra quant bias

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

inter quant bias

-color_table_id[:stream_specifier] integer ()

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

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

Possible values:

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

context model

-slice_flags[:stream_specifier] integer ()

-xvmc_acceleration[:stream_specifier] integer ()

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

macroblock decision algorithm (high quality mode)

 

Possible values:

-stream_codec_tag[:stream_specifier] integer ()

-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

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

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

Possible values:

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

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

deprecated, use mpegvideo private options instead

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

Possible values:

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

motion estimation threshold

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

macroblock threshold

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

intra_dc_precision

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

nsse weight

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

number of macroblock rows at the top which are skipped

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

number of macroblock rows at the bottom which are skipped

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

Possible values:

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

Possible values:

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

frame skip threshold

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

frame skip factor

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

frame skip exponent

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

frame skip compare function

 

Possible values:

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

increase the quantizer for macroblocks close to borders

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

minimum macroblock Lagrange factor (VBR)

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

maximum macroblock Lagrange factor (VBR)

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

motion estimation bitrate penalty compensation (1.0 = 256)

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

Possible values:

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

Possible values:

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

Possible values:

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

refine the two motion vectors used in bidirectional macroblocks

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

downscale frames for dynamic B-frame decision

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

minimum interval between IDR-frames (x264)

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

reference frames to consider for motion compensation

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

chroma QP offset from luma

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

rate-distortion optimal quantization

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

multiplied by qscale for each frame and added to scene_change_score

-mv0_threshold[:stream_specifier] integer (output,video)

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

adjust sensitivity of b_frame_strategy 1

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

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

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

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

GOP timecode frame start number, in non-drop-frame format

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

set desired number of audio channels

-bits_per_raw_sample[:stream_specifier] integer ()

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

Possible values:

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

Possible values:

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

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

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

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

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

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

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

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

-log_level_offset[:stream_specifier] integer ()

set the log level offset

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

number of slices, used in parallelized encoding

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

select multithreading type

 

Possible values:

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

audio service type

 

Possible values:

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

Possible values:

Format AVOptions

-probesize integer (input)

set probing size

-packetsize integer (output)

set packet size

-fflags flags (input/output)

Possible values:

-analyzeduration integer (input)

how many microseconds are analyzed to estimate duration

-cryptokey hexadecimal string (input)

decryption key

-indexmem integer (input)

max memory used for timestamp index (per stream)

-rtbufsize integer (input)

max memory used for buffering real-time frames

-fdebug flags (input/output)

print specific debug info

 

Possible values:

-max_delay integer (input/output)

maximum muxing or demuxing delay in microseconds

-fpsprobesize integer (input)

number of frames used to probe fps

-f_err_detect flags (input)

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

 

Possible values:

-err_detect flags (input)

set error detection flags

 

Possible values:

Main options

-f format

Force format to use.

-of formatter

Use a specific formatter to output the document. The following formatters are available

-unit

Show the unit of the displayed values.

-prefix

Use SI prefixes for the displayed values. Unless the "-byte_binary_prefix" option is used all the prefixes are decimal.

-byte_binary_prefix

Force the use of binary prefixes for byte values.

-sexagesimal

Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.

-pretty

Prettify the format of the displayed values, it corresponds to the options "-unit -prefix -byte_binary_prefix -sexagesimal".

-show_format

Show information about the container format of the input multimedia stream.

 

All the container format information is printed within a section with name "FORMAT".

-show_format_entry name

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_packets

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_streams

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".

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 "%0Nd", 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 "%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 "%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".

crc

CRC (Cyclic Redundancy Check) testing format. This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a single line of the form: CRC=0x CRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames. For example to compute the CRC of the input, and store it in the file out.crc:
avconv -i INPUT -f crc out.crc
You can print the CRC to stdout with the command:
avconv -i INPUT -f crc -
You can select the output format of each frame with avconv by specifying the audio and video codec and format. For example to compute the CRC of the input audio converted to PCM unsigned 8-bit and the input video converted to MPEG-2 video, use the command:
avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
See also the framecrc muxer.

framecrc

Per-frame CRC (Cyclic Redundancy Check) testing format. This muxer computes and prints the Adler-32 CRC for each decoded audio and video frame. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a line for each audio and video frame of the form: stream_index, frame_dts, frame_size, 0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the decoded frame. For example to compute the CRC of each decoded frame in the input, and store it in the file out.crc:
avconv -i INPUT -f framecrc out.crc
You can print the CRC of each decoded frame to stdout with the command:
avconv -i INPUT -f framecrc -
You can select the output format of each frame with avconv by specifying the audio and video codec and format. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command:
avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
See also the crc muxer.

hls

Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming specification. It creates a playlist file and numbered segment files. The output filename specifies the playlist filename; the segment filenames receive the same basename as the playlist, a sequential number and a .ts extension.
avconv -i in.nut out.m3u8

-hls_time seconds

Set the segment length in seconds.

-hls_list_size size

Set the maximum number of playlist entries.

-hls_wrap wrap

Set the number after which index wraps.

-start_number number

Start the sequence from number.

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 "%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 "%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.

MOV/MP4/ISMV

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

-movflags frag_keyframe

Start a new fragment at each video keyframe.

-frag_duration duration

Create fragments that are duration microseconds long.

-frag_size size

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

-movflags frag_custom

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

-min_frag_duration duration

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

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

-movflags empty_moov

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

 

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

-movflags separate_moof

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

 

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

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)

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).

The recognized metadata settings in mpegts muxer are "service_provider" and "service_name". If they are not set the default for "service_provider" is "Libav" and the default for "service_name" is "Service01".
avconv -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts

null

Null muxer. This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes. For example to benchmark decoding with avconv you can use the command:
avconv -benchmark -i INPUT -f null out.null
Note that the above command does not read or write the out.null file, but specifying the output file is required by the avconv syntax. Alternatively you can write the command as:
avconv -benchmark -i INPUT -f null -

matroska

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

title=title name

Name provided to a single track

language=language name

Specifies the language of the track in the Matroska languages form

STEREO_MODE=mode

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

For example a 3D WebM clip can be created using the following command line:
avconv -i sample_left_right_clip.mpg -an -c:v libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webm

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

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

concat

Physical concatenation protocol. Allow to read and seek from many resource in sequence as if they were a unique resource. A URL accepted by this protocol has the syntax:
concat:<URL1>|<URL2>|...|<URLN>
where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol. For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with avplay use the command:
avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
Note that you may need to escape the character "|" which is special for many shells.

file

File access protocol. Allow to read from or read to a file. For example to read from a file input.mpeg with avconv use the command:
avconv -i file:input.mpeg output.mpeg
The av* tools default to the file protocol, that is a resource specified with the name "FILE.mpeg" is interpreted as the URL "file:FILE.mpeg".

gopher

Gopher protocol.

hls

Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+ proto" after the hls URI scheme name, where proto is either "file" or "http".
hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8
Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files.

http

HTTP (Hyper Text Transfer Protocol).

mmst

MMS (Microsoft Media Server) protocol over TCP.

mmsh

MMS (Microsoft Media Server) protocol over HTTP. The required syntax is:
mmsh://<server>[:<port>][/<app>][/<playpath>]

md5

MD5 output protocol. Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file. Some examples follow.
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
avconv -i input.flv -f avi -y md5:output.avi.md5

# Write the MD5 hash of the encoded AVI file to stdout.
avconv -i input.flv -f avi -y md5:
Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.

pipe

UNIX pipe access protocol. Allow to read and write from UNIX pipes. The accepted syntax is:
pipe:[<number>]
number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. For example to read from stdin with avconv:
cat test.wav | avconv -i pipe:0
# ...this is the same as...
cat test.wav | avconv -i pipe:
For writing to stdout with avconv:
avconv -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
avconv -i test.wav -f avi pipe: | cat > test.avi
Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.

rtmp

Real-Time Messaging Protocol. The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network. The required syntax is:
rtmp://<server>[:<port>][/<app>][/<instance>][/<playpath>]
The accepted parameters are:

server

The address of the RTMP server.

port

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

app

It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You can override the value parsed from the URI through the "rtmp_app" option, too.

playpath

It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by "mp4:". You can override the value parsed from the URI through the "rtmp_playpath" option, too.

listen

Act as a server, listening for an incoming connection.

timeout

Maximum time to wait for the incoming connection. Implies listen.

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

rtmp_app

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

rtmp_buffer

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

rtmp_conn

Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each value is prefixed by a single character denoting the type, B for Boolean, N for number, S for string, O for object, or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively. Data items in subobjects may be named, by prefixing the type with 'N' and specifying the name before the value (i.e. "NB:myFlag:1"). This option may be used multiple times to construct arbitrary AMF sequences.

rtmp_flashver

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

rtmp_flush_interval

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

rtmp_live

Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is "any", which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are "live" and "recorded".

rtmp_pageurl

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

rtmp_playpath

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

rtmp_subscribe

Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.

rtmp_swfhash

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

rtmp_swfsize

Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl

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

rtmp_swfverify

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

rtmp_tcurl

URL of the target stream. Defaults to proto://host[:port]/app.

For example to read with avplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver":
avplay rtmp://myserver/vod/sample

rtmpe

Encrypted Real-Time Messaging Protocol. The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a pair of RC4 keys.

rtmps

Real-Time Messaging Protocol over a secure SSL connection. The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted connection.

rtmpt

Real-Time Messaging Protocol tunneled through HTTP. The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content within HTTP requests to traverse firewalls.

rtmpte

Encrypted Real-Time Messaging Protocol tunneled through HTTP. The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming multimedia content within HTTP requests to traverse firewalls.

rtmpts

Real-Time Messaging Protocol tunneled through HTTPS. The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls.

rtmp, rtmpe, rtmps, rtmpt, rtmpte

Real-Time Messaging Protocol and its variants supported through librtmp. Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with "--enable-librtmp". If enabled this will replace the native RTMP protocol. This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS). The required syntax is:
<rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key=val. See the librtmp manual page (man 3 librtmp) for more information. For example, to stream a file in real-time to an RTMP server using avconv:
avconv -re -i myfile -f flv rtmp://myserver/live/mystream
To play the same stream using avplay:
avplay "rtmp://myserver/live/mystream live=1"

rtp

Real-Time Protocol.

rtsp

RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT). The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
RTSP server ("http://github.com/revmischa/rtsp-server")). The required syntax for a RTSP url is:
rtsp://<hostname>[:<port>]/<path>
The following options (set on the avconv/avplay command line, or set in code via "AVOption"s or in "avformat_open_input"), are supported: Flags for "rtsp_transport":

udp

Use UDP as lower transport protocol.

tcp

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

udp_multicast

Use UDP multicast as lower transport protocol.

http

Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.

Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). For the muxer, only the "tcp" and "udp" options are supported. Flags for "rtsp_flags":

filter_src

Accept packets only from negotiated peer address and port.

listen

Act as a server, listening for an incoming connection.

When receiving data over UDP, the demuxer tries to reorder received packets (since they may arrive out of order, or packets may get lost totally). This can be disabled by setting the maximum demuxing delay to zero (via the "max_delay" field of AVFormatContext). When watching multi-bitrate Real-RTSP streams with avplay, the streams to display can be chosen with "-vst" n and "-ast" n for video and audio respectively, and can be switched on the fly by pressing "v" and "a". Example command lines: To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
To watch a stream tunneled over HTTP:
avplay -rtsp_transport http rtsp://server/video.mp4
To send a stream in realtime to a RTSP server, for others to watch:
avconv -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
To receive a stream in realtime:
avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

sap

Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port. Muxer The syntax for a SAP url given to the muxer is:
sap://<destination>[:<port>][?<options>]
The RTP packets are sent to destination on port port, or to port 5004 if no port is specified. options is a "&"-separated list. The following options are supported:

announce_addr=address

Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.

announce_port=port

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

ttl=ttl

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

same_port=0|1

If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.

Example command lines follow. To broadcast a stream on the local subnet, for watching in VLC:
avconv -re -i <input> -f sap sap://224.0.0.255?same_port=1
Similarly, for watching in avplay:
avconv -re -i <input> -f sap sap://224.0.0.255
And for watching in avplay, over IPv6:
avconv -re -i <input> -f sap sap://[ff0e::1:2:3:4]
Demuxer The syntax for a SAP url given to the demuxer is:
sap://[<address>][:<port>]
address is the multicast address to listen for announcements on, if omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted. The demuxers listens for announcements on the given address and port. Once an announcement is received, it tries to receive that particular stream. Example command lines follow. To play back the first stream announced on the normal SAP multicast address:
avplay sap://
To play back the first stream announced on one the default IPv6 SAP multicast address:
avplay sap://[ff0e::2:7ffe]

tcp

Trasmission Control Protocol. The required syntax for a TCP url is:
tcp://<hostname>:<port>[?<options>]

listen

Listen for an incoming connection

 

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

udp

User Datagram Protocol. The required syntax for a UDP url is:
udp://<hostname>:<port>[?<options>]
options contains a list of &-seperated options of the form key=val. Follow the list of supported options.

buffer_size=size

set the UDP buffer size in bytes

localport=port

override the local UDP port to bind with

localaddr=addr

Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface.

pkt_size=size

set the size in bytes of UDP packets

reuse=1|0

explicitly allow or disallow reusing UDP sockets

ttl=ttl

set the time to live value (for multicast only)

connect=1|0

Initialize the UDP socket with "connect()". In this case, the destination address can't be changed with ff_udp_set_remote_url later. If the destination address isn't known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port.

sources=address[,address]

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

block=address[,address]

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

Some usage examples of the udp protocol with avconv follow. To stream over UDP to a remote endpoint:
avconv -i <input> -f <format> udp://<hostname>:<port>
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
avconv -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
To receive over UDP from a remote endpoint:
avconv -i udp://[<multicast-address>]:<port>

alsa

ALSA (Advanced Linux Sound Architecture) input device. To enable this input device during configuration you need libasound installed on your system. This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier. An ALSA identifier has the syntax:
hw:<CARD>[,<DEV>[,<SUBDEV>]]
where the DEV and SUBDEV components are optional. The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and subdevice number (-1 means any). To see the list of cards currently recognized by your system check the files /proc/asound/cards and /proc/asound/devices. For example to capture with avconv from an ALSA device with card id 0, you may run the command:
avconv -f alsa -i hw:0 alsaout.wav
For more information see: < http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html>

bktr

BSD video input device.

dv1394

Linux DV 1394 input device.

fbdev

Linux framebuffer input device. The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually /dev/fb0. For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source tree. To record from the framebuffer device /dev/fb0 with avconv:
avconv -f fbdev -r 10 -i /dev/fb0 out.avi
You can take a single screenshot image with the command:
avconv -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

jack

JACK input device. To enable this input device during configuration you need libjack installed on your system. A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the Libav input device. Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients. To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect programs, or do it through a graphical interface, for example with qjackctl. To list the JACK clients and their properties you can invoke the command jack_lsp. Follows an example which shows how to capture a JACK readable client with avconv.
# Create a JACK writable client with name "libav".
$ avconv -f jack -i libav -y out.wav

# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000

# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
libav:input_1
metro:120_bpm

# Connect metro to the avconv writable client.
$ jack_connect metro:120_bpm libav:input_1
For more information read: < http://jackaudio.org/>

libdc1394

IIDC1394 input device, based on libdc1394 and libraw1394.

oss

Open Sound System input device. The filename to provide to the input device is the device node representing the OSS input device, and is usually set to /dev/dsp. For example to grab from /dev/dsp using avconv use the command:
avconv -f oss -i /dev/dsp /tmp/oss.wav
For more information about OSS see: < http://manuals.opensound.com/usersguide/dsp.html>

pulse

pulseaudio input device. To enable this input device during configuration you need libpulse-simple installed in your system. The filename to provide to the input device is a source device or the string "default" To list the pulse source devices and their properties you can invoke the command pactl list sources.
avconv -f pulse -i default /tmp/pulse.wav
server AVOption The syntax is:
-server <server name>
Connects to a specific server. name AVOption The syntax is:
-name <application name>
Specify the application name pulse will use when showing active clients, by default it is "libav" stream_name AVOption The syntax is:
-stream_name <stream name>
Specify the stream name pulse will use when showing active streams, by default it is "record" sample_rate AVOption The syntax is:
-sample_rate <samplerate>
Specify the samplerate in Hz, by default 48kHz is used. channels AVOption The syntax is:
-channels <N>
Specify the channels in use, by default 2 (stereo) is set. frame_size AVOption The syntax is:
-frame_size <bytes>
Specify the number of byte per frame, by default it is set to 1024. fragment_size AVOption The syntax is:
-fragment_size <bytes>
Specify the minimal buffering fragment in pulseaudio, it will affect the audio latency. By default it is unset.

sndio

sndio input device. To enable this input device during configuration you need libsndio installed on your system. The filename to provide to the input device is the device node representing the sndio input device, and is usually set to /dev/audio0. For example to grab from /dev/audio0 using avconv use the command:
avconv -f sndio -i /dev/audio0 /tmp/oss.wav

video4linux2

Video4Linux2 input video device. The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the kind /dev/videoN, where N is a number associated to the device. Video4Linux2 devices usually support a limited set of widthxheight sizes and framerates. You can check which are supported using -list_formats all for Video4Linux2 devices. Some usage examples of the video4linux2 devices with avconv and avplay:
# Grab and show the input of a video4linux2 device.
avplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

# Grab and record the input of a video4linux2 device, leave the
framerate and size as previously set.
avconv -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

vfwcap

VfW (Video for Windows) capture input device. The filename passed as input is the capture driver number, ranging from 0 to 9. You may use "list" as filename to print a list of drivers. Any other filename will be interpreted as device number 0.

x11grab

X11 video input device. This device allows to capture a region of an X11 display. The filename passed as input has the syntax:
[<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]
hostname:display_number.screen_number specifies the X11 display name of the screen to grab from. hostname can be omitted, and defaults to "localhost". The environment variable DISPLAY contains the default display name. x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the X11 screen. They default to 0. Check the X11 documentation (e.g. man X) for more detailed information. Use the dpyinfo program for getting basic information about the properties of your X11 display (e.g. grep for "name" or "dimensions"). For example to grab from :0.0 using avconv:
avconv -f x11grab -r 25 -s cif -i :0.0 out.mpg

# Grab at position 10,20.
avconv -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
follow_mouse AVOption The syntax is:
-follow_mouse centered|<PIXELS>
When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within PIXELS (greater than zero) to the edge of region. For example:
avconv -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg

# Follows only when the mouse pointer reaches within 100 pixels to edge
avconv -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
show_region AVOption The syntax is:
-show_region 1
If show_region AVOption is specified with 1, then the grabbing region will be indicated on screen. With this option, it's easy to know what is being grabbed if only a portion of the screen is grabbed. For example:
avconv -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg

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