The generic syntax is:
ffprobe [options] [‘input_file’] |
ffprobe 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, ffprobe 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.
ffprobe 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 ffprobe or for specifying which information to display, and for setting how ffprobe will show it.
ffprobe output is designed to be easily parsable by a textual filter, and consists of one or more sections of a form defined by the selected writer, which is specified by the ‘print_format’ option.
Metadata tags stored in the container or in the streams are recognized and printed in the corresponding "FORMAT" or "STREAM" section.
All the numerical options, if not specified otherwise, accept in input a string representing a number, which may contain one of the International System number postfixes, for example ’K’, ’M’, ’G’. If ’i’ is appended after the postfix, powers of 2 are used instead of powers of 10. The ’B’ postfix multiplies the value for 8, and can be appended after another postfix or used alone. This allows using for example ’KB’, ’MiB’, ’G’ and ’B’ as postfix.
Options which do not take arguments are boolean options, and set the corresponding value to true. They can be set to false by prefixing with "no" the option name, for example using "-nofoo" in the command line will set to false the boolean option with name "foo".
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:
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.
These options are shared amongst the av* tools.
Show license.
Show help.
Show version.
Show available formats.
The fields preceding the format names have the following meanings:
Decoding available
Encoding available
Show available codecs.
The fields preceding the codec names have the following meanings:
Decoding available
Encoding available
Video/audio/subtitle codec
Codec supports slices
Codec supports direct rendering
Codec can handle input truncated at random locations instead of only at frame boundaries
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 FFmpeg version.
Dump full command line and console output to a file named
program-YYYYMMDD-HHMMSS.log in the current
directory.
This file can be useful for bug reports.
It also implies -loglevel verbose.
Note: setting the environment variable FFREPORT to any value has the
same effect.
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:
ffmpeg -i input.flac -id3v2_version 3 out.mp3 |
All codec AVOptions are obviously per-stream, so the chapter on stream specifiers applies to them
Note ‘-nooption’ syntax cannot be used for boolean AVOptions, use ‘-option 0’/‘-option 1’.
Note2 old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon.
Force format to use.
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".
Set the output printing format.
writer_name specifies the name of the writer, and writer_options specifies the options to be passed to the writer.
For example for printing the output in JSON format, specify:
-print_format json |
For more details on the available output printing formats, see the Writers section below.
Show information about the error found when trying to probe the input.
The error information is printed within a section with name "ERROR".
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 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 frame contained in the input multimedia stream.
The information for each single frame is printed within a dedicated section with name "FRAME".
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".
Show private data, that is data depending on the format of the particular shown element. This option is enabled by default, but you may need to disable it for specific uses, for example when creating XSD-compliant XML output.
Show information related to program version.
Version information is printed within a section with name "PROGRAM_VERSION".
Show information related to library versions.
Version information for each library is printed within a section with name "LIBRARY_VERSION".
Show information related to program and library versions. This is the equivalent of setting both ‘-show_program_version’ and ‘-show_library_versions’ options.
Read input_file.
A writer defines the output format adopted by ffprobe, and will be
used for printing all the parts of the output.
A writer may accept one or more arguments, which specify the options to adopt.
A description of the currently available writers follows.
Default format.
Print each section in the form:
[SECTION] key1=val1 ... keyN=valN [/SECTION] |
Metadata tags are printed as a line in the corresponding FORMAT or STREAM section, and are prefixed by the string "TAG:".
Compact format.
Each section is printed on a single line. If no option is specifid, the output has the form:
section|key1=val1| ... |keyN=valN |
Metadata tags are printed in the corresponding "format" or "stream" section. A metadata tag key, if printed, is prefixed by the string "tag:".
This writer accepts options as a list of key=value pairs, separated by ":".
The description of the accepted options follows.
Specify the character to use for separating fields in the output line. It must be a single printable character, it is "|" by default.
If set to 1 specify not to print the key of each field. Its default value is 0.
Set the escape mode to use, default to "c".
It can assume one of the following values:
Perform C-like escaping. Strings containing a newline (’\n’) or carriage return (’\r’), the escaping character (’\’) or the item separator character SEP are escaped using C-like fashioned escaping, so that a newline is converted to the sequence "\n", a carriage return to "\r", ’\’ to "\\" and the separator SEP is converted to "\SEP".
Perform CSV-like escaping, as described in RFC4180. Strings containing a newline (’\n’), a carriage return (’\r’), a double quote (’"’), or SEP are enclosed in double-quotes.
Perform no escaping.
CSV format.
This writer is equivalent to
compact=item_sep=,:nokey=1:escape=csv.
JSON based format.
Each section is printed using JSON notation.
This writer accepts options as a list of key=value pairs, separated by ":".
The description of the accepted options follows.
If set to 1 enable compact output, that is each section will be printed on a single line. Default value is 0.
For more information about JSON, see http://www.json.org/.
XML based format.
The XML output is described in the XML schema description file ‘ffprobe.xsd’ installed in the FFmpeg datadir.
Note that the output issued will be compliant to the ‘ffprobe.xsd’ schema only when no special global output options (‘unit’, ‘prefix’, ‘byte_binary_prefix’, ‘sexagesimal’ etc.) are specified.
This writer accepts options as a list of key=value pairs, separated by ":".
The description of the accepted options follows.
If set to 1 specify if the output should be fully qualified. Default value is 0. This is required for generating an XML file which can be validated through an XSD file.
If set to 1 perform more checks for ensuring that the output is XSD compliant. Default value is 0. This option automatically sets ‘fully_qualified’ to 1.
For more information about the XML format, see http://www.w3.org/XML/.
ffprobe supports Timecode extraction:
Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding --enable-lib option. You can list all
available decoders using the configure option --list-decoders.
You can disable all the decoders with the configure option
--disable-decoders and selectively enable / disable single decoders
with the options --enable-decoder=DECODER /
--disable-decoder=DECODER.
The option -codecs of the ff* tools will display the list of
enabled decoders.
A description of some of the currently available video decoders follows.
Raw video decoder.
This decoder decodes rawvideo streams.
Specify the assumed field type of the input video.
the video is assumed to be progressive (default)
bottom-field-first is assumed
top-field-first is assumed
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.
Demuxers are configured elements in FFmpeg which allow to read the multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all available ones using the configure option "–list-demuxers".
You can disable all the demuxers using the configure option "–disable-demuxers", and selectively enable a single demuxer with the option "–enable-demuxer=DEMUXER", or disable it with the option "–disable-demuxer=DEMUXER".
The option "-formats" of the ff* tools will display the list of enabled demuxers.
The description of some of the currently available demuxers follows.
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 ffmpeg for creating a
video from the images in the file sequence ‘img-001.jpeg’,
‘img-002.jpeg’, ..., assuming an input frame rate of 10 frames per
second:
ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv |
Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to convert a single image file ‘img.jpeg’ you can employ the command:
ffmpeg -i img.jpeg img.png |
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting the discard flags on AVStreams (by pressing ’a’ or ’v’ in ffplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named "variant_bitrate".
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen http://uazu.net/sbagen/ to generate binaural beats sessions. A SBG script looks like that:
-SE a: 300-2.5/3 440+4.5/0 b: 300-2.5/0 440+4.5/3 off: - NOW == a +0:07:00 == b +0:14:00 == a +0:21:00 == b +0:30:00 off |
A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps (including the script start time) or only relative ones, then its layout is fixed, and the conversion is straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference for relative timestamps will be taken from the current time of day at the time the script is read, and the script layout will be frozen according to that reference. That means that if the script is directly played, the actual times will match the absolute timestamps up to the sound controller’s clock accuracy, but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.
Protocols are configured elements in FFmpeg which allow to access resources which require the use of a particular protocol.
When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "–list-protocols".
You can disable all the protocols using the configure option "–disable-protocols", and selectively enable a protocol using the option "–enable-protocol=PROTOCOL", or you can disable a particular protocol using the option "–disable-protocol=PROTOCOL".
The option "-protocols" of the ff* tools will display the list of supported protocols.
A description of the currently available protocols follows.
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were a unique resource.
A URL accepted by this protocol has the syntax:
concat:URL1|URL2|...|URLN |
where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol.
For example to read a sequence of files ‘split1.mpeg’,
‘split2.mpeg’, ‘split3.mpeg’ with ffplay use the
command:
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg |
Note that you may need to escape the character "|" which is special for many shells.
File access protocol.
Allow to read from or read to a file.
For example to read from a file ‘input.mpeg’ with ffmpeg
use the command:
ffmpeg -i file:input.mpeg output.mpeg |
The ff* tools default to the file protocol, that is a resource specified with the name "FILE.mpeg" is interpreted as the URL "file:FILE.mpeg".
Gopher 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).
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. ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5: |
Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
pipe:[number] |
number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading.
For example to read from stdin with ffmpeg:
cat test.wav | ffmpeg -i pipe:0 # ...this is the same as... cat test.wav | ffmpeg -i pipe: |
For writing to stdout with ffmpeg:
ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... ffmpeg -i test.wav -f avi pipe: | cat > test.avi |
Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.
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][/playpath] |
The accepted parameters are:
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.).
It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by "mp4:".
For example to read with ffplay a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
ffplay rtmp://myserver/vod/sample |
Real-Time Messaging Protocol and its variants supported through librtmp.
Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with "–enable-librtmp". If enabled this will replace the native RTMP protocol.
This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
rtmp_proto://server[:port][/app][/playpath] options |
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key=val.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
ffmpeg:
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream |
To play the same stream using ffplay:
ffplay "rtmp://myserver/live/mystream live=1" |
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 ffmpeg/ffplay 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.
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). In
order for this to be enabled, a maximum delay must be specified in the
max_delay field of AVFormatContext.
When watching multi-bitrate Real-RTSP streams with ffplay, the
streams to display can be chosen with -vst n and
-ast n for video and audio respectively, and can be switched
on the fly by pressing v and a.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 |
To watch a stream tunneled over HTTP:
ffplay -rtsp_transport http rtsp://server/video.mp4 |
To send a stream in realtime to a RTSP server, for others to watch:
ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp |
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:
ffmpeg -re -i input -f sap sap://224.0.0.255?same_port=1 |
Similarly, for watching in ffplay:
ffmpeg -re -i input -f sap sap://224.0.0.255 |
And for watching in ffplay, over IPv6:
ffmpeg -re -i input -f sap sap://[ff0e::1:2:3:4] |
The syntax for a SAP url given to the demuxer is:
sap://[address][:port] |
address is the multicast address to listen for announcements on, if omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port. Once an announcement is received, it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
ffplay sap:// |
To play back the first stream announced on one the default IPv6 SAP multicast address:
ffplay sap://[ff0e::2:7ffe] |
Trasmission Control Protocol.
The required syntax for a TCP url is:
tcp://hostname:port[?options] |
Listen for an incoming connection
ffmpeg -i input -f format tcp://hostname:port?listen ffplay tcp://hostname:port |
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.
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.
Some usage examples of the udp protocol with ffmpeg follow.
To stream over UDP to a remote endpoint:
ffmpeg -i input -f format udp://hostname:port |
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
ffmpeg -i input -f mpegts udp://hostname:port?pkt_size=188&buffer_size=65535 |
To receive over UDP from a remote endpoint:
ffmpeg -i udp://[multicast-address]:port |
Input devices are configured elements in FFmpeg which allow to access the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices are enabled by default. You can list all available ones using the configure option "–list-indevs".
You can disable all the input devices using the configure option "–disable-indevs", and selectively enable an input device using the option "–enable-indev=INDEV", or you can disable a particular input device using the option "–disable-indev=INDEV".
The option "-formats" of the ff* tools will display the list of supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound installed on your system.
This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
hw:CARD[,DEV[,SUBDEV]] |
where the DEV and SUBDEV components are optional.
The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and subdevice number (-1 means any).
To see the list of cards currently recognized by your system check the files ‘/proc/asound/cards’ and ‘/proc/asound/devices’.
For example to capture with ffmpeg from an ALSA device with
card id 0, you may run the command:
ffmpeg -f alsa -i hw:0 alsaout.wav |
For more information see: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html
BSD video input device.
Windows DirectShow input device.
DirectShow support is enabled when FFmpeg is built with mingw-w64. Currently only audio and video devices are supported.
Multiple devices may be opened as separate inputs, but they may also be opened on the same input, which should improve synchronism between them.
The input name should be in the format:
TYPE=NAME[:TYPE=NAME] |
where TYPE can be either audio or video, and NAME is the device’s name.
If no options are specified, the device’s defaults are used. If the device does not support the requested options, it will fail to open.
Set the video size in the captured video.
Set the framerate in the captured video.
Set the sample rate (in Hz) of the captured audio.
Set the sample size (in bits) of the captured audio.
Set the number of channels in the captured audio.
If set to ‘true’, print a list of devices and exit.
If set to ‘true’, print a list of selected device’s options and exit.
Set video device number for devices with same name (starts at 0, defaults to 0).
Set audio device number for devices with same name (starts at 0, defaults to 0).
$ ffmpeg -list_devices true -f dshow -i dummy |
$ ffmpeg -f dshow -i video="Camera" |
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera" |
$ ffmpeg -f dshow -i video="Camera":audio="Microphone" |
$ ffmpeg -list_options true -f dshow -i video="Camera" |
Linux DV 1394 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
ffmpeg:
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi |
You can take a single screenshot image with the command:
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg |
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
JACK input device.
To enable this input device during configuration you need libjack installed on your system.
A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.
Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the jack_connect
and jack_disconnect programs, or do it through a graphical interface,
for example with qjackctl.
To list the JACK clients and their properties you can invoke the command
jack_lsp.
Follows an example which shows how to capture a JACK readable client
with ffmpeg.
# Create a JACK writable client with name "ffmpeg". $ ffmpeg -f jack -i ffmpeg -y out.wav # Start the sample jack_metro readable client. $ jack_metro -b 120 -d 0.2 -f 4000 # List the current JACK clients. $ jack_lsp -c system:capture_1 system:capture_2 system:playback_1 system:playback_2 ffmpeg:input_1 metro:120_bpm # Connect metro to the ffmpeg writable client. $ jack_connect metro:120_bpm ffmpeg:input_1 |
For more information read: http://jackaudio.org/
Libavfilter input virtual device.
This input device reads data from the open output pads of a libavfilter filtergraph.
For each filtergraph open output, the input device will create a corresponding stream which is mapped to the generated output. Currently only video data is supported. The filtergraph is specified through the option ‘graph’.
Specify the filtergraph to use as input. Each video open output must be labelled by a unique string of the form "outN", where N is a number starting from 0 corresponding to the mapped input stream generated by the device. The first unlabelled output is automatically assigned to the "out0" label, but all the others need to be specified explicitly.
If not specified defaults to the filename specified for the input device.
ffplay:
ffplay -f lavfi -graph "color=pink [out0]" dummy |
ffplay -f lavfi color=pink |
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 |
ffplay:
ffplay -f lavfi "amovie=test.wav" |
ffplay:
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" |
IIDC1394 input device, based on libdc1394 and libraw1394.
The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1 implementation.
To enable this input device during configuration, you need OpenAL
headers and libraries installed on your system, and need to configure
FFmpeg with --enable-openal.
OpenAL headers and libraries should be provided as part of your OpenAL
implementation, or as an additional download (an SDK). Depending on your
installation you may need to specify additional flags via the
--extra-cflags and --extra-ldflags for allowing the build
system to locate the OpenAL headers and libraries.
An incomplete list of OpenAL implementations follows:
The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See http://openal.org/.
Portable, open source (LGPL) software implementation. Includes backends for the most common sound APIs on the Windows, Linux, Solaris, and BSD operating systems. See http://kcat.strangesoft.net/openal.html.
OpenAL is part of Core Audio, the official Mac OS X Audio interface. See http://developer.apple.com/technologies/mac/audio-and-video.html
This device allows to capture from an audio input device handled through OpenAL.
You need to specify the name of the device to capture in the provided filename. If the empty string is provided, the device will automatically select the default device. You can get the list of the supported devices by using the option list_devices.
Set the number of channels in the captured audio. Only the values ‘1’ (monaural) and ‘2’ (stereo) are currently supported. Defaults to ‘2’.
Set the sample size (in bits) of the captured audio. Only the values ‘8’ and ‘16’ are currently supported. Defaults to ‘16’.
Set the sample rate (in Hz) of the captured audio. Defaults to ‘44.1k’.
If set to ‘true’, print a list of devices and exit. Defaults to ‘false’.
Print the list of OpenAL supported devices and exit:
$ ffmpeg -list_devices true -f openal -i dummy out.ogg |
Capture from the OpenAL device ‘DR-BT101 via PulseAudio’:
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg |
Capture from the default device (note the empty string ” as filename):
$ ffmpeg -f openal -i '' out.ogg |
Capture from two devices simultaneously, writing to two different files,
within the same ffmpeg command:
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg |
Note: not all OpenAL implementations support multiple simultaneous capture - try the latest OpenAL Soft if the above does not work.
Open Sound System input device.
The filename to provide to the input device is the device node representing the OSS input device, and is usually set to ‘/dev/dsp’.
For example to grab from ‘/dev/dsp’ using ffmpeg use the
command:
ffmpeg -f oss -i /dev/dsp /tmp/oss.wav |
For more information about OSS see: http://manuals.opensound.com/usersguide/dsp.html
pulseaudio input device.
To enable this input device during configuration you need libpulse-simple installed in your system.
The filename to provide to the input device is a source device or the string "default"
To list the pulse source devices and their properties you can invoke
the command pactl list sources.
ffmpeg -f pulse -i default /tmp/pulse.wav |
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 the LIBAVFORMAT_IDENT string
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 ffmpeg use the
command:
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav |
Video4Linux and Video4Linux2 input video devices.
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.
Video4Linux and Video4Linux2 devices only support a limited set of
widthxheight sizes and framerates. You can check which are
supported for example with the command dov4l for Video4Linux
devices and using -list_formats all for Video4Linux2 devices.
If the size for the device is set to 0x0, the input device will try to auto-detect the size to use. Only for the video4linux2 device, if the frame rate is set to 0/0 the input device will use the frame rate value already set in the driver.
Video4Linux support is deprecated since Linux 2.6.30, and will be dropped in later versions.
Note that if FFmpeg is build with v4l-utils support ("–enable-libv4l2" option), it will always be used.
Follow some usage examples of the video4linux devices with the ff* tools.
# Grab and show the input of a video4linux device, frame rate is set # to the default of 25/1. ffplay -s 320x240 -f video4linux /dev/video0 # Grab and show the input of a video4linux2 device, auto-adjust size. ffplay -f video4linux2 /dev/video0 # Grab and record the input of a video4linux2 device, auto-adjust size, # frame rate value defaults to 0/0 so it is read from the video4linux2 # driver. ffmpeg -f video4linux2 -i /dev/video0 out.mpeg |
"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and "video4linux2".
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 ffmpeg:
ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg # Grab at position 10,20. ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg |
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:
ffmpeg -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 ffmpeg -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:
ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg # With follow_mouse ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg |