FFmpeg Documentation
The generic syntax is:
ffmpeg [[infile options][@option{-i} infile]]... {[outfile options] outfile}...
FFmpeg is a very fast video and audio converter. It can also grab from a live audio/video source.
The command line interface is designed to be intuitive, in the sense that FFmpeg tries to figure out all parameters that can possibly be derived automatically. You usually only have to specify the target bitrate you want.
FFmpeg can also convert from any sample rate to any other, and resize video on the fly with a high quality polyphase filter.
As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file.
* To set the video bitrate of the output file to 64kbit/s:
ffmpeg -i input.avi -b 64k output.avi
* To force the frame rate of the output file to 24 fps:
ffmpeg -i input.avi -r 24 output.avi
* To force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps:
ffmpeg -r 1 -i input.m2v -r 24 output.avi
The format option may be needed for raw input files.
By default, FFmpeg tries to convert as losslessly as possible: It uses the same audio and video parameters for the outputs as the one specified for the inputs.
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 commandline will set to false the boolean option with name "foo".
These options are shared amongst the ff* tools.
hh:mm:ss[.xxx]
syntax is also supported.
hh:mm:ss[.xxx]
syntax is also supported.
[-]hh:mm:ss[.xxx]
syntax is also supported.
This option affects all the input files that follow it.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by 'offset' seconds.
now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z])If the value is "now" it takes the current time. Time is local time unless 'Z' or 'z' is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day.
ffmpeg -i in.avi -metadata title="my title" out.flv
ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpgNevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2vIt is of little use elsewise.
copy
special value to
tell that the raw codec data must be copied as is.
ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
tex^qComp
).
ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
copy
special value to
specify that the raw codec data must be copied as is.
-newaudio
(-acodec
, -ab
, etc..).
Mapping will be done automatically, if the number of output streams is equal to
the number of input streams, else it will pick the first one that matches. You
can override the mapping using -map
as usual.
Example:
ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which would be awkward to specify on the command line. Lines starting with the hash ('#') character are ignored and are used to provide comments. Check the `ffpresets' directory in the FFmpeg source tree for examples.
Preset files are specified with the vpre
, apre
,
spre
, and fpre
options. The fpre
option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the vpre
, apre
, and
spre
options, the options specified in a preset file are
applied to the currently selected codec of the same type as the preset
option.
The argument passed to the vpre
, apre
, and spre
preset options identifies the preset file to use according to the
following rules:
First ffmpeg searches for a file named arg.ffpreset in the
directories `$FFMPEG_DATADIR' (if set), and `$HOME/.ffmpeg', and in
the datadir defined at configuration time (usually `PREFIX/share/ffmpeg')
in that order. For example, if the argument is libx264-max
, it will
search for the file `libx264-max.ffpreset'.
If no such file is found, then ffmpeg will search for a file named
codec_name-arg.ffpreset in the above-mentioned
directories, where codec_name is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with -vcodec libx264
and use -vpre max
,
then it will search for the file `libx264-max.ffpreset'.
@anchor{FFmpeg formula evaluator}
When evaluating a rate control string, FFmpeg uses an internal formula evaluator.
The following binary operators are available: +
, -
,
*
, /
, ^
.
The following unary operators are available: +
, -
,
(...)
.
The following statements are available: ld
, st
,
while
.
The following functions are available:
The following constants are available:
ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
FFmpeg can grab video and audio from devices given that you specify the input format and device.
ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
Note that you must activate the right video source and channel before launching FFmpeg with any TV viewer such as xawtv (http://linux.bytesex.org/xawtv/) by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.
FFmpeg can grab the X11 display.
ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.
ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x-offset and 20 the y-offset for the grabbing.
* FFmpeg can use any supported file format and protocol as input:
Examples:
* You can use YUV files as input:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the @option{-s} option if FFmpeg cannot guess it.
* You can input from a raw YUV420P file:
ffmpeg -i /tmp/test.yuv /tmp/out.avi
test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.
* You can output to a raw YUV420P file:
ffmpeg -i mydivx.avi hugefile.yuv
* You can set several input files and output files:
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.
* You can also do audio and video conversions at the same time:
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
Converts a.wav to MPEG audio at 22050 Hz sample rate.
* You can encode to several formats at the same time and define a mapping from input stream to output streams:
ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map file:index' specifies which input stream is used for each output stream, in the order of the definition of output streams.
* You can transcode decrypted VOBs:
ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
This is a typical DVD ripping example; the input is a VOB file, the
output an AVI file with MPEG-4 video and MP3 audio. Note that in this
command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
input video. Furthermore, the audio stream is MP3-encoded so you need
to enable LAME support by passing --enable-libmp3lame
to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use ffmpeg -formats
.
* You can extract images from a video, or create a video from many images:
For extracting images from a video:
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
This will extract one video frame per second from the video and will output them in files named `foo-001.jpeg', `foo-002.jpeg', etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the above command in combination with the -vframes or -t option, or in combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
The syntax foo-%03d.jpeg
specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
* You can put many streams of the same type in the output:
ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
In addition to the first video and audio streams, the resulting output file `test12.avi' will contain the second video and the second audio stream found in the input streams list.
The -newvideo
, -newaudio
and -newsubtitle
options have to be specified immediately after the name of the output
file to which you want to add them.
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.
Linux DV 1394 input device.
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/
IIDC1394 input device, based on libdc1394 and libraw1394.
Open Sound System input device.
The filename to provide to the input device is the device node representing the OSS input device, and is usually set to `/dev/dsp'.
For example to grab from `/dev/dsp' using `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
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 the command `v4l-info' for Video4Linux2 devices.
If the size for the device is set to 0x0, the input device will try to autodetect the size to use.
Video4Linux support is deprecated since Linux 2.6.30, and will be dropped in later versions.
Follow some usage examples of the video4linux devices with the ff* tools.
# Grab and show the input of a video4linux device. ffplay -s 320x240 -f video4linux /dev/video0 # Grab and show the input of a video4linux2 device, autoadjust size. ffplay -f video4linux2 /dev/video0 # Grab and record the input of a video4linux2 device, autoadjust size. ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
VfW (Video for Windows) capture input device.
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 ommitted, and defaults to "localhost". The environment variable @env{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 -25 -s cif -i :0.0+10,20 out.mpg
Output devices are configured elements in FFmpeg which allow to write multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices are enabled by default. You can list all available ones using the configure option "--list-outdevs".
You can disable all the output devices using the configure option "--disable-outdevs", and selectively enable an output device using the option "--enable-outdev=OUTDEV", or you can disable a particular input device using the option "--disable-outdev=OUTDEV".
The option "-formats" of the ff* tools will display the list of enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
ALSA (Advanced Linux Sound Architecture) output device.
OSS (Open Sound System) output device.
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.
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 multimeā dia content across a TCP/IP network.
The required syntax is:
rtmp://server[:port][/app][/playpath]
The accepted parameters are:
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 explicitely 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.
Trasmission Control Protocol.
User Datagram Protocol.
When you configure your FFmpeg build, you can disable any of the existing filters using --disable-filters. The configure output will show the audio filters included in your build.
Below is a description of the currently available audio filters.
Pass the audio source unchanged to the output.
When you configure your FFmpeg build, you can disable any of the existing filters using --disable-filters. The configure output will show the video filters included in your build.
Below is a description of the currently available video filters.
Crop the input video to x:y:width:height.
./ffmpeg -i in.avi -vf "crop=0:0:0:240" out.avi
x and y specify the position of the top-left corner of the output (non-cropped) area.
The default value of x and y is 0.
The width and height parameters specify the width and height of the output (non-cropped) area.
A value of 0 is interpreted as the maximum possible size contained in the area delimited by the top-left corner at position x:y.
For example the parameters:
"crop=100:100:0:0"
will delimit the rectangle with the top-left corner placed at position 100:100 and the right-bottom corner corresponding to the right-bottom corner of the input image.
The default value of width and height is 0.
Buffer input images and send them when they are requested.
This filter is mainly useful when auto-inserted by the libavfilter framework.
The filter does not take parameters.
Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is supported for the input to the next filter.
The filter accepts a list of pixel format names, separated by ":", for example "yuv420p:monow:rgb24".
The following command:
./ffmpeg -i in.avi -vf "format=yuv420p" out.avi
will convert the input video to the format "yuv420p".
Flip the input video horizontally.
For example to horizontally flip the video in input with `ffmpeg':
ffmpeg -i in.avi -vf "hflip" out.avi
Force libavfilter not to use any of the specified pixel formats for the input to the next filter.
The filter accepts a list of pixel format names, separated by ":", for example "yuv420p:monow:rgb24".
The following command:
./ffmpeg -i in.avi -vf "noformat=yuv420p, vflip" out.avi
will make libavfilter use a format different from "yuv420p" for the input to the vflip filter.
Pass the video source unchanged to the output.
Apply smooth transform using libopencv.
To enable this filter install libopencv library and headers and configure FFmpeg with --enable-libopencv.
It accepts the following parameters: type:param1:param2:param3:param4.
type is the type of smooth filter to apply, and can be one of the following value: "blur", "blur_no_scale", "median", "gaussian", "bilateral". The default value is "gaussian".
param1, param2, param3, and param4 are parameters whose meanings depend on smooth type. param1 and param2 accept integer positive values or 0, param3 and param4 accept float values.
The default value for param1 is 3, the default value for the other parameters is 0.
These parameters corresponds to the parameters assigned to the
libopencv function cvSmooth
. Refer the official libopencv
documentation for the exact meaning of the parameters:
http://opencv.willowgarage.com/documentation/c/image_filtering.html
Add paddings to the input image, and places the original input at the given coordinates x, y.
It accepts the following parameters: width:height:x:y:color.
Follows the description of the accepted parameters.
Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video.
For example:
format=monow, pixdesctest
can be used to test the monowhite pixel format descriptor definition.
Scale the input video to width:height and/or convert the image format.
For example the command:
./ffmpeg -i in.avi -vf "scale=200:100" out.avi
will scale the input video to a size of 200x100.
If the input image format is different from the format requested by the next filter, the scale filter will convert the input to the requested format.
If the value for width or height is 0, the respective input size is used for the output.
If the value for width or height is -1, the scale filter will use, for the respective output size, a value that maintains the aspect ratio of the input image.
The default value of width and height is 0.
Pass the images of input video on to next video filter as multiple slices.
./ffmpeg -i in.avi -vf "slicify=32" out.avi
The filter accepts the slice height as parameter. If the parameter is not specified it will use the default value of 16.
Adding this in the beginning of filter chains should make filtering faster due to better use of the memory cache.
Sharpen or blur the input video.
It accepts the following parameters: luma_msize_x:luma_msize_y:luma_amount:chroma_msize_x:chroma_msize_y:chroma_amount
Negative values for the amount will blur the input video, while positive values will sharpen. All parameters are optional and default to the equivalent of the string '5:5:1.0:0:0:0.0'.
# Strong luma sharpen effect parameters unsharp=7:7:2.5 # Strong blur of both luma and chroma parameters unsharp=7:7:-2:7:7:-2 # Use the default values with @command{ffmpeg} ./ffmpeg -i in.avi -vf "unsharp" out.mp4
Flip the input video vertically.
./ffmpeg -i in.avi -vf "vflip" out.avi
Below is a description of the currently available video sources.
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in `libavfilter/vsrc_buffer.h'.
It accepts the following parameters: width:height:pix_fmt_string
All the parameters need to be explicitely defined.
Follows the list of the accepted parameters.
For example:
buffer=320:240:yuv410p
will instruct the source to accept video frames with size 320x240 and with format "yuv410p". Since the pixel format with name "yuv410p" corresponds to the number 6 (check the enum PixelFormat definition in `libavutil/pixfmt.h'), this example corresponds to:
buffer=320:240:6
Provide an uniformly colored input.
It accepts the following parameters: color:frame_size:frame_rate
Follows the description of the accepted parameters.
For example the following graph description will generate a red source with an opacity of 0.2, with size "qcif" and a frame rate of 10 frames per second, which will be overlayed over the source connected to the pad with identifier "in".
"color=red@0.2:qcif:10 [color]; [in][color] overlay [out]"
Null video source, never return images. It is mainly useful as a template and to be employed in analysis / debugging tools.
It accepts as optional parameter a string of the form width:height, where width and height specify the size of the configured source.
The default values of width and height are respectively 352 and 288 (corresponding to the CIF size format).
Below is a description of the currently available video sinks.
Null video sink, do absolutely nothing with the input video. It is mainly useful as a template and to be employed in analysis / debugging tools.
This document was generated on 15 September 2010 using texi2html 1.56k.