-segment_time_delta 0.05 out%03d.nut In order to force key frames on the input file, transcoding is required. · To convert the in.mkv to TS segments using the "libx264" and "libfaac" encoders: ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts · Segment the input file, and create an M3U8 live playlist (can be used as live HLS source): ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \ -segment_list_flags +live -segment_time 10 out%03d.mkv 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: ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 To attach a picture to an mp3 file select both the audio and the picture stream with "map": ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 INPUT DEVICES 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 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> bktr BSD video input device. dshow Windows DirectShow input device. DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. 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. Options 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. video_size Set the video size in the captured video. framerate Set the framerate in the captured video. sample_rate Set the sample rate (in Hz) of the captured audio. sample_size Set the sample size (in bits) of the captured audio. channels Set the number of channels in the captured audio. list_devices If set to true, print a list of devices and exit. list_options If set to true, print a list of selected device's options and exit. video_device_number Set video device number for devices with same name (starts at 0, defaults to 0). audio_device_number Set audio device number for devices with same name (starts at 0, defaults to 0). pixel_format Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo. audio_buffer_size Set audio device buffer size in milliseconds (which can directly impact latency, depending on the device). Defaults to using the audio device's default buffer size (typically some multiple of 500ms). Setting this value too low can degrade performance. See also <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx> Examples · Print the list of DirectShow supported devices and exit: $ ffmpeg -list_devices true -f dshow -i dummy · Open video device Camera: $ ffmpeg -f dshow -i video="Camera" · Open second video device with name Camera: $ ffmpeg -f dshow -video_device_number 1 -i video="Camera" · Open video device Camera and audio device Microphone: $ ffmpeg -f dshow -i video="Camera":audio="Microphone" · Print the list of supported options in selected device and exit: $ ffmpeg -list_options true -f dshow -i video="Camera" 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 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). iec61883 FireWire DV/HDV input device using libiec61883. To enable this input device, you need libiec61883, libraw1394 and libavc1394 installed on your system. Use the configure option "--enable-libiec61883" to compile with the device enabled. The iec61883 capture device supports capturing from a video device connected via IEEE1394 (FireWire), using libiec61883 and the new Linux FireWire stack (juju). This is the default DV/HDV input method in Linux Kernel 2.6.37 and later, since the old FireWire stack was removed. Specify the FireWire port to be used as input file, or "auto" to choose the first port connected. Options dvtype Override autodetection of DV/HDV. This should only be used if auto detection does not work, or if usage of a different device type should be prohibited. Treating a DV device as HDV (or vice versa) will not work and result in undefined behavior. The values auto, dv and hdv are supported. dvbuffer Set maxiumum size of buffer for incoming data, in frames. For DV, this is an exact value. For HDV, it is not frame exact, since HDV does not have a fixed frame size. Examples · Grab and show the input of a FireWire DV/HDV device. ffplay -f iec61883 -i auto · Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000 packets if the source is HDV. ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg 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 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/> lavfi 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. Options 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. Examples · Create a color video stream and play it back with ffplay: ffplay -f lavfi -graph "color=pink [out0]" dummy · As the previous example, but use filename for specifying the graph description, and omit the "out0" label: ffplay -f lavfi color=pink · Create three different video test filtered sources and play them: ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 · Read an audio stream from a file using the amovie source and play it back with ffplay: ffplay -f lavfi "amovie=test.wav" · Read an audio stream and a video stream and play it back with ffplay: ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" libdc1394 IIDC1394 input device, based on libdc1394 and libraw1394. openal 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: Creative The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See <http://openal.org/>. OpenAL Soft 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>. Apple 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. Options channels Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are currently supported. Defaults to 2. sample_size Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently supported. Defaults to 16. sample_rate Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k. list_devices If set to true, print a list of devices and exit. Defaults to false. Examples 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. 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 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> 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. ffmpeg -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 the LIBAVFORMAT_IDENT string 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 ffmpeg use the command: ffmpeg -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 ffmpeg and ffplay: The time base for the timestamps is 1 microsecond. Depending on the kernel version and configuration, the timestamps may be derived from the real time clock (origin at the Unix Epoch) or the monotonic clock (origin usually at boot time, unaffected by NTP or manual changes to the clock). The -timestamps abs or -ts abs option can be used to force conversion into the real time clock. Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2" option), it will always be used. # Grab and show the input of a video4linux2 device. ffplay -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. ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg "v4l" and "v4l2" can be used as aliases for the respective "video4linux" and "video4linux2". 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 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 Options draw_mouse Specify whether to draw the mouse pointer. A value of 0 specify not to draw the pointer. Default value is 1. follow_mouse Make the grabbed area follow the mouse. The argument can be "centered" or a number of pixels 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 To follow 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 framerate Set the grabbing frame rate. Default value is "ntsc", corresponding to a framerate of "30000/1001". show_region Show grabbed region on screen. If show_region is specified with 1, then the grabbing region will be indicated on screen. With this option, it is 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 video_size Set the video frame size. Default value is "vga". OUTPUT DEVICES 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 ALSA (Advanced Linux Sound Architecture) output device. caca CACA output device. This output devices allows to show a video stream in CACA window. Only one CACA window is allowed per application, so you can have only one instance of this output device in an application. To enable this output device you need to configure FFmpeg with "--enable-libcaca". libcaca is a graphics library that outputs text instead of pixels. For more information about libcaca, check: <http://caca.zoy.org/wiki/libcaca> Options window_title Set the CACA window title, if not specified default to the filename specified for the output device. window_size Set the CACA window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video. driver Set display driver. algorithm Set dithering algorithm. Dithering is necessary because the picture being rendered has usually far more colours than the available palette. The accepted values are listed with "-list_dither algorithms". antialias Set antialias method. Antialiasing smoothens the rendered image and avoids the commonly seen staircase effect. The accepted values are listed with "-list_dither antialiases". charset Set which characters are going to be used when rendering text. The accepted values are listed with "-list_dither charsets". color Set color to be used when rendering text. The accepted values are listed with "-list_dither colors". list_drivers If set to true, print a list of available drivers and exit. list_dither List available dither options related to the argument. The argument must be one of "algorithms", "antialiases", "charsets", "colors". Examples · The following command shows the ffmpeg output is an CACA window, forcing its size to 80x25: ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca - · Show the list of available drivers and exit: ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true - · Show the list of available dither colors and exit: ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - oss OSS (Open Sound System) output device. sdl SDL (Simple DirectMedia Layer) output device. This output devices allows to show a video stream in an SDL window. Only one SDL window is allowed per application, so you can have only one instance of this output device in an application. To enable this output device you need libsdl installed on your system when configuring your build. For more information about SDL, check: <http://www.libsdl.org/> Options window_title Set the SDL window title, if not specified default to the filename specified for the output device. icon_title Set the name of the iconified SDL window, if not specified it is set to the same value of window_title. window_size Set the SDL window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video, downscaled according to the aspect ratio. Examples The following command shows the ffmpeg output is an SDL window, forcing its size to the qcif format: ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output" sndio sndio audio output device. PROTOCOLS 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. bluray Read BluRay playlist. The accepted options are: angle BluRay angle chapter Start chapter (1...N) playlist Playlist to read (BDMV/PLAYLIST/?????.mpls) Examples: Read longest playlist from BluRay mounted to /mnt/bluray: bluray:/mnt/bluray Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray 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 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 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 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. 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. 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 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. 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 ffplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver": ffplay 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 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" 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 <http://github.com/revmischa/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 "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 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 To receive a stream in realtime: ffmpeg -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 <IP_ADDRESS> (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: ffmpeg -re -i <input> -f sap sap://<IP_ADDRESS>?same_port=1 Similarly, for watching in ffplay: ffmpeg -re -i <input> -f sap sap://<IP_ADDRESS> And for watching in ffplay, over IPv6: ffmpeg -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 <IP_ADDRESS> (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] tcp Trasmission Control Protocol. The required syntax for a TCP url is: tcp://<hostname>:<port>[?<options>] listen Listen for an incoming connection timeout=microseconds In read mode: if no data arrived in more than this time interval, raise error. In write mode: if socket cannot be written in more than this time interval, raise error. This also sets timeout on TCP connection establishing. ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen ffplay tcp://<hostname>:<port> tls Transport Layer Security/Secure Sockets Layer The required syntax for a TLS/SSL url is: tls://<hostname>:<port>[?<options>] listen Act as a server, listening for an incoming connection. cafile=filename Certificate authority file. The file must be in OpenSSL PEM format. cert=filename Certificate file. The file must be in OpenSSL PEM format. key=filename Private key file. verify=0|1 Verify the peer's certificate. Example command lines: To create a TLS/SSL server that serves an input stream. ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key> To play back a stream from the TLS/SSL server using ffplay: ffplay tls://<hostname>:<port> udp User Datagram Protocol. The required syntax for a UDP url is: udp://<hostname>:<port>[?<options>] options contains a list of &-separated options of the form key=val. In case threading is enabled on the system, a circular buffer is used to store the incoming data, which allows to reduce loss of data due to UDP socket buffer overruns. The fifo_size and overrun_nonfatal options are related to this buffer. The list of supported options follows. buffer_size=size Set the UDP socket buffer size in bytes. This is used both for the receiving and the sending buffer size. 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. fifo_size=units Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096. overrun_nonfatal=1|0 Survive in case of UDP receiving circular buffer overrun. Default value is 0. timeout=microseconds In read mode: if no data arrived in more than this time interval, raise error. 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> BITSTREAM FILTERS When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option "--list-bsfs". You can disable all the bitstream filters using the configure option "--disable-bsfs", and selectively enable any bitstream filter using the option "--enable-bsf=BSF", or you can disable a particular bitstream filter using the option "--disable-bsf=BSF". The option "-bsfs" of the ff* tools will display the list of all the supported bitstream filters included in your build. Below is a description of the currently available bitstream filters. aac_adtstoasc chomp dump_extradata h264_mp4toannexb Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification). This is required by some streaming formats, typically the MPEG-2 transport stream format ("mpegts"). For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the command: ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts imx_dump_header mjpeg2jpeg Convert MJPEG/AVI1 packets to full JPEG/JFIF packets. MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>: Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec." This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images. ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg exiftran -i -9 frame*.jpg ffmpeg -i frame_%d.jpg -c:v copy rotated.avi mjpega_dump_header movsub mp3_header_compress mp3_header_decompress noise remove_extradata FILTERING INTRODUCTION Filtering in FFmpeg is enabled through the libavfilter library. Libavfilter is the filtering API of FFmpeg. It is the substitute of the now deprecated 'vhooks' and started as a Google Summer of Code project. Audio filtering integration into the main FFmpeg repository is a work in progress, so audio API and ABI should not be considered stable yet. In libavfilter, it is possible for filters to have multiple inputs and multiple outputs. To illustrate the sorts of things that are possible, we can use a complex filter graph. For example, the following one: input --> split ---------------------> overlay --> output | ^ | | +-----> crop --> vflip -------+ splits the stream in two streams, sends one stream through the crop filter and the vflip filter before merging it back with the other stream by overlaying it on top. You can use the following command to achieve this: ffmpeg -i input -vf "[in] split [T1], [T2] overlay=0:H/2 [out]; [T1] crop=iw:ih/2:0:ih/2, vflip [T2]" output The result will be that in output the top half of the video is mirrored onto the bottom half. Filters are loaded using the -vf or -af option passed to ffmpeg or to ffplay. Filters in the same linear chain are separated by commas. In our example, split, overlay are in one linear chain, and crop, vflip are in another. The points where the linear chains join are labeled by names enclosed in square brackets. In our example, that is [T1] and [T2]. The special labels [in] and [out] are the points where video is input and output. Some filters take in input a list of parameters: they are specified after the filter name and an equal sign, and are separated from each other by a colon. There exist so-called source filters that do not have an audio/video input, and sink filters that will not have audio/video output. GRAPH The graph2dot program included in the FFmpeg tools directory can be used to parse a filter graph description and issue a corresponding textual representation in the dot language. Invoke the command: graph2dot -h to see how to use graph2dot. You can then pass the dot description to the dot program (from the graphviz suite of programs) and obtain a graphical representation of the filter graph. For example the sequence of commands: echo <GRAPH_DESCRIPTION> | \ tools/graph2dot -o graph.tmp && \ dot -Tpng graph.tmp -o graph.png && \ display graph.png can be used to create and display an image representing the graph described by the GRAPH_DESCRIPTION string. Note that this string must be a complete self-contained graph, with its inputs and outputs explicitly defined. For example if your command line is of the form: ffmpeg -i infile -vf scale=640:360 outfile your GRAPH_DESCRIPTION string will need to be of the form: nullsrc,scale=640:360,nullsink you may also need to set the nullsrc parameters and add a format filter in order to simulate a specific input file. FILTERGRAPH DESCRIPTION A filtergraph is a directed graph of connected filters. It can contain cycles, and there can be multiple links between a pair of filters. Each link has one input pad on one side connecting it to one filter from which it takes its input, and one output pad on the other side connecting it to the one filter accepting its output. Each filter in a filtergraph is an instance of a filter class registered in the application, which defines the features and the number of input and output pads of the filter.  Filtergraph syntax A filtergraph can be represented using a textual representation, which is recognized by the -filter/-vf and -filter_complex options in ffmpeg and -vf in ffplay, and by the "avfilter_graph_parse()"/"avfilter_graph_parse2()" function defined in libavfilter/avfiltergraph.h. A filterchain consists of a sequence of connected filters, each one connected to the previous one in the sequence. A filterchain is represented by a list of ","-separated filter descriptions. A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list of ";"-separated filterchain descriptions. A filter is represented by a string of the form: [in_link_1]...[in_link_N]filter_name=arguments[out_link_1]...[out_link_M] filter_name is the name of the filter class of which the described filter is an instance of, and has to be the name of one of the filter classes registered in the program. The name of the filter class is optionally followed by a string "=arguments". arguments is a string which contains the parameters used to initialize the filter instance, and are described in the filter descriptions below. The list of arguments can be quoted using the character "'" as initial and ending mark, and the character '\' for escaping the characters within the quoted text; otherwise the argument string is considered terminated when the next special character (belonging to the set "[]=;,") is encountered. The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads. When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created. If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain: nullsrc, split[L1], [L2]overlay, nullsink the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled. In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected. Libavfilter will automatically insert scale filters where format conversion is required. It is possible to specify swscale flags for those automatically inserted scalers by prepending "sws_flags=flags;" to the filtergraph description. Follows a BNF description for the filtergraph syntax: <NAME> ::= sequence of alphanumeric characters and '_' <LINKLABEL> ::= "[" <NAME> "]" <LINKLABELS> ::= <LINKLABEL> [<LINKLABELS>] <FILTER_ARGUMENTS> ::= sequence of chars (eventually quoted) <FILTER> ::= [<LINKNAMES>] <NAME> ["=" <ARGUMENTS>] [<LINKNAMES>] <FILTERCHAIN> ::= <FILTER> [,<FILTERCHAIN>] <FILTERGRAPH> ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>] AUDIO FILTERS 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. aconvert Convert the input audio format to the specified formats. The filter accepts a string of the form: "sample_format:channel_layout". sample_format specifies the sample format, and can be a string or the corresponding numeric value defined in libavutil/samplefmt.h. Use 'p' suffix for a planar sample format. channel_layout specifies the channel layout, and can be a string or the corresponding number value defined in libavutil/audioconvert.h. The special parameter "auto", signifies that the filter will automatically select the output format depending on the output filter. Some examples follow. · Convert input to float, planar, stereo: aconvert=fltp:stereo · Convert input to unsigned 8-bit, automatically select out channel layout: aconvert=u8:auto aformat Convert the input audio to one of the specified formats. The framework will negotiate the most appropriate format to minimize conversions. The filter accepts the following named parameters: sample_fmts A comma-separated list of requested sample formats. sample_rates A comma-separated list of requested sample rates. channel_layouts A comma-separated list of requested channel layouts. If a parameter is omitted, all values are allowed. For example to force the output to either unsigned 8-bit or signed 16-bit stereo: aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo amerge Merge two or more audio streams into a single multi-channel stream. The filter accepts the following named options: inputs Set the number of inputs. Default is 2. If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the output will be set accordingly and the channels will be reordered as necessary. If the channel layouts of the inputs are not disjoint, the output will have all the channels of the first input then all the channels of the second input, in that order, and the channel layout of the output will be the default value corresponding to the total number of channels. For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then the output will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the first input, b1 is the first channel of the second input). On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or may not be the expected value. All inputs must have the same sample rate, and format. If inputs do not have the same duration, the output will stop with the shortest. Example: merge two mono files into a stereo stream: amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge Example: multiple merges: ffmpeg -f lavfi -i " amovie=input.mkv:si=0 [a0]; amovie=input.mkv:si=1 [a1]; amovie=input.mkv:si=2 [a2]; amovie=input.mkv:si=3 [a3]; amovie=input.mkv:si=4 [a4]; amovie=input.mkv:si=5 [a5]; [a0][a1][a2][a3][a4][a5] amerge=inputs=6" -c:a pcm_s16le output.mkv amix Mixes multiple audio inputs into a single output. For example ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds. The filter accepts the following named parameters: inputs Number of inputs. If unspecified, it defaults to 2. duration How to determine the end-of-stream. longest Duration of longest input. (default) shortest Duration of shortest input. first Duration of first input. dropout_transition Transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds. anull Pass the audio source unchanged to the output. aresample Resample the input audio to the specified sample rate. The filter accepts exactly one parameter, the output sample rate. If not specified then the filter will automatically convert between its input and output sample rates. For example, to resample the input audio to 44100Hz: aresample=44100 asetnsamples Set the number of samples per each output audio frame. The last output packet may contain a different number of samples, as the filter will flush all the remaining samples when the input audio signal its end. The filter accepts parameters as a list of key=value pairs, separated by ":". nb_out_samples, n Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. Default value is 1024. pad, p If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will contain the same number of samples as the previous ones. Default value is 1. For example, to set the number of per-frame samples to 1234 and disable padding for the last frame, use: asetnsamples=n=1234:p=0 ashowinfo Show a line containing various information for each input audio frame. The input audio is not modified. The shown line contains a sequence of key/value pairs of the form key:value. A description of each shown parameter follows: n sequential number of the input frame, starting from 0 pts presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad, and is usually 1/sample_rate. pts_time presentation TimeStamp of the input frame, expressed as a number of seconds pos position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic audio) fmt sample format name chlayout channel layout description nb_samples number of samples (per each channel) contained in the filtered frame rate sample rate for the audio frame checksum Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame plane_checksum Adler-32 checksum (printed in hexadecimal) for each input frame plane, expressed in the form "[c0 c1 c2 c3 c4 c5 c6 c7]" asplit Split input audio into several identical outputs. The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2. For example: [in] asplit [out0][out1] will create two separate outputs from the same input. To create 3 or more outputs, you need to specify the number of outputs, like in: [in] asplit=3 [out0][out1][out2] ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT will create 5 copies of the input audio. astreamsync Forward two audio streams and control the order the buffers are forwarded. The argument to the filter is an expression deciding which stream should be forwarded next: if the result is negative, the first stream is forwarded; if the result is positive or zero, the second stream is forwarded. It can use the following variables: b1 b2 number of buffers forwarded so far on each stream s1 s2 number of samples forwarded so far on each stream t1 t2 current timestamp of each stream The default value is "t1-t2", which means to always forward the stream that has a smaller timestamp. Example: stress-test "amerge" by randomly sending buffers on the wrong input, while avoiding too much of a desynchronization: amovie=file.ogg [a] ; amovie=file.mp3 [b] ; [a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ; [a2] [b2] amerge atempo Adjust audio tempo. The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 2.0] range. For example, to slow down audio to 80% tempo: atempo=0.8 For example, to speed up audio to 125% tempo: atempo=1.25 earwax Make audio easier to listen to on headphones. This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers). Ported from SoX. pan Mix channels with specific gain levels. The filter accepts the output channel layout followed by a set of channels definitions. This filter is also designed to remap efficiently the channels of an audio stream. The filter accepts parameters of the form: "l:outdef:outdef:..." l output channel layout or number of channels outdef output channel specification, of the form: "out_name=[gain*]in_name[+[gain*]in_name...]" out_name output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.) gain multiplicative coefficient for the channel, 1 leaving the volume unchanged in_name input channel to use, see out_name for details; it is not possible to mix named and numbered input channels If the `=' in a channel specification is replaced by `<', then the gains for that specification will be renormalized so that the total is 1, thus avoiding clipping noise. Mixing examples For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel: pan=1:c0=0.9*c0+0.1*c1 A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround: pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR Note that ffmpeg integrates a default down-mix (and up-mix) system that should be preferred (see "-ac" option) unless you have very specific needs. Remapping examples The channel remapping will be effective if, and only if: *<gain coefficients are zeroes or ones,> *<only one input per channel output,> If all these conditions are satisfied, the filter will notify the user ("Pure channel mapping detected"), and use an optimized and lossless method to do the remapping. For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra channels: pan="stereo: c0=FL : c1=FR" Given the same source, you can also switch front left and front right channels and keep the input channel layout: pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5" If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with: pan="stereo:c1=c1" Still with a stereo audio stream input, you can copy the right channel in both front left and right: pan="stereo: c0=FR : c1=FR" silencedetect Detect silence in an audio stream. This filter logs a message when it detects that the input audio volume is less or equal to a noise tolerance value for a duration greater or equal to the minimum detected noise duration. The printed times and duration are expressed in seconds. duration, d Set silence duration until notification (default is 2 seconds). noise, n Set noise tolerance. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default is -60dB, or 0.001. Detect 5 seconds of silence with -50dB noise tolerance: silencedetect=n=-50dB:d=5 Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in silence.mp3: ffmpeg -f lavfi -i amovie=silence.mp3,silencedetect=noise=0.0001 -f null - volume Adjust the input audio volume. The filter accepts exactly one parameter vol, which expresses how the audio volume will be increased or decreased. Output values are clipped to the maximum value. If vol is expressed as a decimal number, the output audio volume is given by the relation: <output_volume> = <vol> * <input_volume> If vol is expressed as a decimal number followed by the string "dB", the value represents the requested change in decibels of the input audio power, and the output audio volume is given by the relation: <output_volume> = 10^(<vol>/20) * <input_volume> Otherwise vol is considered an expression and its evaluated value is used for computing the output audio volume according to the first relation. Default value for vol is 1.0. Examples · Half the input audio volume: volume=0.5 The above example is equivalent to: volume=1/2 · Decrease input audio power by 12 decibels: volume=-12dB volumedetect Detect the volume of the input video. The filter has no parameters. The input is not modified. Statistics about the volume will be printed in the log when the input stream end is reached. In particular it will show the mean volume (root mean square), maximum volume (on a per-sample basis), and the beginning of an histogram of the registered volume values (from the maximum value to a cumulated 1/1000 of the samples). All volumes are in decibels relative to the maximum PCM value. Here is an excerpt of the output: [Parsed_volumedetect_0 0xa23120] mean_volume: -27 dB [Parsed_volumedetect_0 0xa23120] max_volume: -4 dB [Parsed_volumedetect_0 0xa23120] histogram_4db: 6 [Parsed_volumedetect_0 0xa23120] histogram_5db: 62 [Parsed_volumedetect_0 0xa23120] histogram_6db: 286 [Parsed_volumedetect_0 0xa23120] histogram_7db: 1042 [Parsed_volumedetect_0 0xa23120] histogram_8db: 2551 [Parsed_volumedetect_0 0xa23120] histogram_9db: 4609 [Parsed_volumedetect_0 0xa23120] histogram_10db: 8409 It means that: · The mean square energy is approximately -27 dB, or 10^-2.7. · The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB. · There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc. In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB causes clipping for 6 samples, etc. asyncts Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed. The filter accepts the following named parameters: compensate Enable stretching/squeezing the data to make it match the timestamps. Disabled by default. When disabled, time gaps are covered with silence. min_delta Minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples. Default value is 0.1. If you get non-perfect sync with this filter, try setting this parameter to 0. max_comp Maximum compensation in samples per second. Relevant only with compensate=1. Default value 500. first_pts Assume the first pts should be this value. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame's expected pts, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with silence if an audio stream starts after the video stream. channelsplit Split each channel in input audio stream into a separate output stream. This filter accepts the following named parameters: channel_layout Channel layout of the input stream. Default is "stereo". For example, assuming a stereo input MP3 file ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv will create an output Matroska file with two audio streams, one containing only the left channel and the other the right channel. To split a 5.1 WAV file into per-channel files ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]' -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]' front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' side_right.wav channelmap Remap input channels to new locations. This filter accepts the following named parameters: channel_layout Channel layout of the output stream. map Map channels from input to output. The argument is a comma-separated list of mappings, each in the "in_channel-out_channel" or in_channel form. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the input channel layout. out_channel is the name of the output channel or its index in the output channel layout. If out_channel is not given then it is implicitly an index, starting with zero and increasing by one for each mapping. If no mapping is present, the filter will implicitly map input channels to output channels preserving index. For example, assuming a 5.1+downmix input MOV file ffmpeg -i in.mov -filter 'channelmap=map=DL-FL\,DR-FR' out.wav will create an output WAV file tagged as stereo from the downmix channels of the input. To fix a 5.1 WAV improperly encoded in AAC's native channel order ffmpeg -i in.wav -filter 'channelmap=1\,2\,0\,5\,3\,4:channel_layout=5.1' out.wav join Join multiple input streams into one multi-channel stream. The filter accepts the following named parameters: inputs Number of input streams. Defaults to 2. channel_layout Desired output channel layout. Defaults to stereo. map Map channels from inputs to output. The argument is a comma-separated list of mappings, each in the "input_idx.in_channel-out_channel" form. input_idx is the 0-based index of the input stream. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the specified input stream. out_channel is the name of the output channel. The filter will attempt to guess the mappings when those are not specified explicitly. It does so by first trying to find an unused matching input channel and if that fails it picks the first unused input channel. E.g. to join 3 inputs (with properly set channel layouts) ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT To build a 5.1 output from 6 single-channel streams: ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex 'join=inputs=6:channel_layout=5.1:map=0.0-FL\,1.0-FR\,2.0-FC\,3.0-SL\,4.0-SR\,5.0-LFE' out resample Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly. AUDIO SOURCES Below is a description of the currently available audio sources. abuffer Buffer audio 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/asrc_abuffer.h. It accepts the following mandatory parameters: sample_rate:sample_fmt:channel_layout sample_rate The sample rate of the incoming audio buffers. sample_fmt The sample format of the incoming audio buffers. Either a sample format name or its corresponging integer representation from the enum AVSampleFormat in libavutil/samplefmt.h channel_layout The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in libavutil/audioconvert.c or its corresponding integer representation from the AV_CH_LAYOUT_* macros in libavutil/audioconvert.h For example: abuffer=44100:s16p:stereo will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with name "s16p" corresponds to the number 6 and the "stereo" channel layout corresponds to the value 0x3, this is equivalent to: abuffer=44100:6:0x3 aevalsrc Generate an audio signal specified by an expression. This source accepts in input one or more expressions (one for each channel), which are evaluated and used to generate a corresponding audio signal. It accepts the syntax: exprs[::options]. exprs is a list of expressions separated by ":", one for each separate channel. In case the channel_layout is not specified, the selected channel layout depends on the number of provided expressions. options is an optional sequence of key=value pairs, separated by ":". The description of the accepted options follows. channel_layout, c Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions. duration, d Set the minimum duration of the sourced audio. See the function "av_parse_time()" for the accepted format. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame. If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. nb_samples, n Set the number of samples per channel per each output frame, default to 1024. sample_rate, s Specify the sample rate, default to 44100. Each expression in exprs can contain the following constants: n number of the evaluated sample, starting from 0 t time of the evaluated sample expressed in seconds, starting from 0 s sample rate Examples · Generate silence: aevalsrc=0 · Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz: aevalsrc="sin(440*2*PI*t)::s=8000" · Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly: aevalsrc="sin(420*2*PI*t):cos(430*2*PI*t)::c=FC|BC" · Generate white noise: aevalsrc="-2+random(0)" · Generate an amplitude modulated signal: aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" · Generate 2.5 Hz binaural beats on a 360 Hz carrier: aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) : 0.1*sin(2*PI*(360+2.5/2)*t)" anullsrc Null audio source, return unprocessed audio frames. It is mainly useful as a template and to be employed in analysis / debugging tools, or as the source for filters which ignore the input data (for example the sox synth filter). It accepts an optional sequence of key=value pairs, separated by ":". The description of the accepted options follows. sample_rate, s Specify the sample rate, and defaults to 44100. channel_layout, cl Specify the channel layout, and can be either an integer or a string representing a channel layout. The default value of channel_layout is "stereo". Check the channel_layout_map definition in libavcodec/audioconvert.c for the mapping between strings and channel layout values. nb_samples, n Set the number of samples per requested frames. Follow some examples: # set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO. anullsrc=r=48000:cl=4 # same as anullsrc=r=48000:cl=mono abuffer Buffer audio frames, and make them available to the filter chain. This source is not intended to be part of user-supplied graph descriptions but for insertion by calling programs through the interface defined in libavfilter/buffersrc.h. It accepts the following named parameters: time_base Timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form. sample_rate Audio sample rate. sample_fmt Name of the sample format, as returned by "av_get_sample_fmt_name()". channel_layout Channel layout of the audio data, in the form that can be accepted by "av_get_channel_layout()". All the parameters need to be explicitly defined. flite Synthesize a voice utterance using the libflite library. To enable compilation of this filter you need to configure FFmpeg with "--enable-libflite". Note that the flite library is not thread-safe. The source accepts parameters as a list of key=value pairs, separated by ":". The description of the accepted parameters follows. list_voices If set to 1, list the names of the available voices and exit immediately. Default value is 0. nb_samples, n Set the maximum number of samples per frame. Default value is 512. textfile Set the filename containing the text to speak. text Set the text to speak. voice, v Set the voice to use for the speech synthesis. Default value is "kal". See also the list_voices option. Examples · Read from file speech.txt, and synthetize the text using the standard flite voice: flite=textfile=speech.txt · Read the specified text selecting the "slt" voice: flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt · Make ffplay speech the specified text, using "flite" and the "lavfi" device: ffplay -f lavfi flite='No more be grieved for which that thou hast done.' For more information about libflite, check: <http://www.speech.cs.cmu.edu/flite/> AUDIO SINKS Below is a description of the currently available audio sinks. abuffersink Buffer audio frames, and make them available to the end of filter chain. This sink is mainly intended for programmatic use, in particular through the interface defined in libavfilter/buffersink.h. It requires a pointer to an AVABufferSinkContext structure, which defines the incoming buffers' formats, to be passed as the opaque parameter to "avfilter_init_filter" for initialization. anullsink Null audio sink, do absolutely nothing with the input audio. It is mainly useful as a template and to be employed in analysis / debugging tools. abuffersink This sink is intended for programmatic use. Frames that arrive on this sink can be retrieved by the calling program using the interface defined in libavfilter/buffersink.h. This filter accepts no parameters. VIDEO FILTERS When you configure your 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. alphaextract Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter. alphamerge Add or replace the alpha component of the primary input with the grayscale value of a second input. This is intended for use with alphaextract to allow the transmission or storage of frame sequences that have alpha in a format that doesn't support an alpha channel. For example, to reconstruct full frames from a normal YUV-encoded video and a separate video created with alphaextract, you might use: movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out] Since this filter is designed for reconstruction, it operates on frame sequences without considering timestamps, and terminates when either input reaches end of stream. This will cause problems if your encoding pipeline drops frames.  To enable compilation of this filter you need to configure FFmpeg with "--enable-libass". This filter accepts the syntax: ass_filename[:options], where ass_filename is the filename of the ASS file to read, and options is an optional sequence of key=value pairs, separated by ":". A description of the accepted options follows. original_size Specifies the size of the original video, the video for which the ASS file was composed. Due to a misdesign in ASS aspect ratio arithmetic, this is necessary to correctly scale the fonts if the aspect ratio has been changed. For example, to render the file sub.ass on top of the input video, use the command: ass=sub.ass bbox Compute the bounding box for the non-black pixels in the input frame luminance plane. This filter computes the bounding box containing all the pixels with a luminance value greater than the minimum allowed value. The parameters describing the bounding box are printed on the filter log. blackdetect Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions, commercials, or invalid recordings. Output lines contains the time for the start, end and duration of the detected black interval expressed in seconds. In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value. This filter accepts a list of options in the form of key=value pairs separated by ":". A description of the accepted options follows. black_min_duration, d Set the minimum detected black duration expressed in seconds. It must be a non-negative floating point number. Default value is 2.0. picture_black_ratio_th, pic_th Set the threshold for considering a picture "black". Express the minimum value for the ratio: <nb_black_pixels> / <nb_pixels> for which a picture is considered black. Default value is 0.98. pixel_black_th, pix_th Set the threshold for considering a pixel "black". The threshold expresses the maximum pixel luminance value for which a pixel is considered "black". The provided value is scaled according to the following equation: <absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size> luminance_range_size and luminance_minimum_value depend on the input video format, the range is [0-255] for YUV full-range formats and [16-235] for YUV non full-range formats. Default value is 0.10. The following example sets the maximum pixel threshold to the minimum value, and detects only black intervals of 2 or more seconds: blackdetect=d=2:pix_th=0.00 blackframe Detect frames that are (almost) completely black. Can be useful to detect chapter transitions or commercials. Output lines consist of the frame number of the detected frame, the percentage of blackness, the position in the file if known or -1 and the timestamp in seconds. In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value. The filter accepts the syntax: blackframe[=<amount>:[<threshold>]] amount is the percentage of the pixels that have to be below the threshold, and defaults to 98. threshold is the threshold below which a pixel value is considered black, and defaults to 32. boxblur Apply boxblur algorithm to the input video. This filter accepts the parameters: luma_radius:luma_power:chroma_radius:chroma_power:alpha_radius:alpha_power Chroma and alpha parameters are optional, if not specified they default to the corresponding values set for luma_radius and luma_power. luma_radius, chroma_radius, and alpha_radius represent the radius in pixels of the box used for blurring the corresponding input plane. They are expressions, and can contain the following constants: w, h the input width and height in pixels cw, ch the input chroma image width and height in pixels hsub, vsub horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1. The radius must be a non-negative number, and must not be greater than the value of the expression "min(w,h)/2" for the luma and alpha planes, and of "min(cw,ch)/2" for the chroma planes. luma_power, chroma_power, and alpha_power represent how many times the boxblur filter is applied to the corresponding plane. Some examples follow: · Apply a boxblur filter with luma, chroma, and alpha radius set to 2: boxblur=2:1 · Set luma radius to 2, alpha and chroma radius to 0 boxblur=2:1:0:0:0:0 · Set luma and chroma radius to a fraction of the video dimension boxblur=min(h\,w)/10:1:min(cw\,ch)/10:1 colormatrix The colormatrix filter allows conversion between any of the following color space: BT.709 (bt709), BT.601 (bt601), SMPTE-240M (smpte240m) and FCC (fcc). The syntax of the parameters is source:destination: colormatrix=bt601:smpte240m copy Copy the input source unchanged to the output. Mainly useful for testing purposes. crop Crop the input video to out_w:out_h:x:y:keep_aspect The keep_aspect parameter is optional, if specified and set to a non-zero value will force the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio. The out_w, out_h, x, y parameters are expressions containing the following constants: x, y the computed values for x and y. They are evaluated for each new frame. in_w, in_h the input width and height iw, ih same as in_w and in_h out_w, out_h the output (cropped) width and height ow, oh same as out_w and out_h a same as iw / ih sar input sample aspect ratio dar input display aspect ratio, it is the same as (iw / ih) * sar hsub, vsub horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1. n the number of input frame, starting from 0 pos the position in the file of the input frame, NAN if unknown t timestamp expressed in seconds, NAN if the input timestamp is unknown The out_w and out_h parameters specify the expressions for the width and height of the output (cropped) video. They are evaluated just at the configuration of the filter. The default value of out_w is "in_w", and the default value of out_h is "in_h". The expression for out_w may depend on the value of out_h, and the expression for out_h may depend on out_w, but they cannot depend on x and y, as x and y are evaluated after out_w and out_h. The x and y parameters specify the expressions for the position of the top-left corner of the output (non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is approximated to the nearest valid value. The default value of x is "(in_w-out_w)/2", and the default value for y is "(in_h-out_h)/2", which set the cropped area at the center of the input image. The expression for x may depend on y, and the expression for y may depend on x. Follow some examples: # crop the central input area with size 100x100 crop=100:100 # crop the central input area with size 2/3 of the input video "crop=2/3*in_w:2/3*in_h" # crop the input video central square crop=in_h # delimit the rectangle with the top-left corner placed at position # 100:100 and the right-bottom corner corresponding to the right-bottom # corner of the input image. crop=in_w-100:in_h-100:100:100 # crop 10 pixels from the left and right borders, and 20 pixels from # the top and bottom borders "crop=in_w-2*10:in_h-2*20" # keep only the bottom right quarter of the input image "crop=in_w/2:in_h/2:in_w/2:in_h/2" # crop height for getting Greek harmony "crop=in_w:1/PHI*in_w" # trembling effect "crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)" # erratic camera effect depending on timestamp "crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" # set x depending on the value of y "crop=in_w/2:in_h/2:y:10+10*sin(n/10)" cropdetect Auto-detect crop size. Calculate necessary cropping parameters and prints the recommended parameters through the logging system. The detected dimensions correspond to the non-black area of the input video. It accepts the syntax: cropdetect[=<limit>[:<round>[:<reset>]]] limit Threshold, which can be optionally specified from nothing (0) to everything (255), defaults to 24. round Value which the width/height should be divisible by, defaults to 16. The offset is automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs. reset Counter that determines after how many frames cropdetect will reset the previously detected largest video area and start over to detect the current optimal crop area. Defaults to 0. This can be useful when channel logos distort the video area. 0 indicates never reset and return the largest area encountered during playback. decimate This filter drops frames that do not differ greatly from the previous frame in order to reduce framerate. The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it could in theory be used for fixing movies that were inverse-telecined incorrectly. It accepts the following parameters: max:hi:lo:frac. max Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped unregarding the number of previous sequentially dropped frames. Default value is 0. hi, lo, frac Set the dropping threshold values. Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out differently over the block. A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of hi, and if no more than frac blocks (1 meaning the whole image) differ by more than a threshold of lo. Default value for hi is 64*12, default value for lo is 64*5, and default value for frac is 0.33. delogo Suppress a TV station logo by a simple interpolation of the surrounding pixels. Just set a rectangle covering the logo and watch it disappear (and sometimes something even uglier appear - your mileage may vary). The filter accepts parameters as a string of the form "x:y:w:h:band", or as a list of key=value pairs, separated by ":". The description of the accepted parameters follows. x, y Specify the top left corner coordinates of the logo. They must be specified. w, h Specify the width and height of the logo to clear. They must be specified. band, t Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 4. show When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, h parameters, and band is set to 4. The default value is 0. Some examples follow. · Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, setting a band of size 10: delogo=0:0:100:77:10 · As the previous example, but use named options: delogo=x=0:y=0:w=100:h=77:band=10 deshake Attempt to fix small changes in horizontal and/or vertical shift. This filter helps remove camera shake from hand-holding a camera, bumping a tripod, moving on a vehicle, etc. The filter accepts parameters as a string of the form "x:y:w:h:rx:ry:edge:blocksize:contrast:search:filename" A description of the accepted parameters follows. x, y, w, h Specify a rectangular area where to limit the search for motion vectors. If desired the search for motion vectors can be limited to a rectangular area of the frame defined by its top left corner, width and height. These parameters have the same meaning as the drawbox filter which can be used to visualise the position of the bounding box. This is useful when simultaneous movement of subjects within the frame might be confused for camera motion by the motion vector search. If any or all of x, y, w and h are set to -1 then the full frame is used. This allows later options to be set without specifying the bounding box for the motion vector search. Default - search the whole frame. rx, ry Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16. edge Specify how to generate pixels to fill blanks at the edge of the frame. An integer from 0 to 3 as follows: 0 Fill zeroes at blank locations 1 Original image at blank locations 2 Extruded edge value at blank locations 3 Mirrored edge at blank locations The default setting is mirror edge at blank locations. blocksize Specify the blocksize to use for motion search. Range 4-128 pixels, default 8. contrast Specify the contrast threshold for blocks. Only blocks with more than the specified contrast (difference between darkest and lightest pixels) will be considered. Range 1-255, default 125. search Specify the search strategy 0 = exhaustive search, 1 = less exhaustive search. Default - exhaustive search. filename If set then a detailed log of the motion search is written to the specified file. drawbox Draw a colored box on the input image. It accepts the syntax: drawbox=<x>:<y>:<width>:<height>:<color> x, y Specify the top left corner coordinates of the box. Default to 0. width, height Specify the width and height of the box, if 0 they are interpreted as the input width and height. Default to 0. color Specify the color of the box to write, it can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence. Follow some examples: # draw a black box around the edge of the input image drawbox # draw a box with color red and an opacity of 50%<EMAIL_ADDRESS>drawtext Draw text string or text from specified file on top of video using the libfreetype library. To enable compilation of this filter you need to configure FFmpeg with "--enable-libfreetype". The filter also recognizes strftime() sequences in the provided text and expands them accordingly. Check the documentation of strftime(). The filter accepts parameters as a list of key=value pairs, separated by ":". The description of the accepted parameters follows. box Used to draw a box around text using background color. Value should be either 1 (enable) or 0 (disable). The default value of box is 0. boxcolor The color to be used for drawing box around text. Either a string (e.g. "yellow") or in 0xRRGGBB[AA] format (e.g. "0xff00ff"), possibly followed by an alpha specifier. The default value of boxcolor is "white". draw Set an expression which specifies if the text should be drawn. If the expression evaluates to 0, the text is not drawn. This is useful for specifying that the text should be drawn only when specific conditions are met. Default value is "1". See below for the list of accepted constants and functions. fix_bounds If true, check and fix text coords to avoid clipping. fontcolor The color to be used for drawing fonts. Either a string (e.g. "red") or in 0xRRGGBB[AA] format (e.g. "0xff000033"), possibly followed by an alpha specifier. The default value of fontcolor is "black". fontfile The font file to be used for drawing text. Path must be included. This parameter is mandatory. fontsize The font size to be used for drawing text. The default value of fontsize is 16. ft_load_flags Flags to be used for loading the fonts. The flags map the corresponding flags supported by libfreetype, and are a combination of the following values: default no_scale no_hinting render no_bitmap vertical_layout force_autohint crop_bitmap pedantic ignore_global_advance_width no_recurse ignore_transform monochrome linear_design no_autohint end table Default value is "render". For more information consult the documentation for the FT_LOAD_* libfreetype flags. shadowcolor The color to be used for drawing a shadow behind the drawn text. It can be a color name (e.g. "yellow") or a string in the 0xRRGGBB[AA] form (e.g. "0xff00ff"), possibly followed by an alpha specifier. The default value of shadowcolor is "black". shadowx, shadowy The x and y offsets for the text shadow position with respect to the position of the text. They can be either positive or negative values. Default value for both is "0". tabsize The size in number of spaces to use for rendering the tab. Default value is 4. timecode Set the initial timecode representation in "hh:mm:ss[:;.]ff" format. It can be used with or without text parameter. timecode_rate option must be specified. timecode_rate, rate, r Set the timecode frame rate (timecode only). text The text string to be drawn. The text must be a sequence of UTF-8 encoded characters. This parameter is mandatory if no file is specified with the parameter textfile. textfile A text file containing text to be drawn. The text must be a sequence of UTF-8 encoded characters. This parameter is mandatory if no text string is specified with the parameter text. If both text and textfile are specified, an error is thrown. x, y The expressions which specify the offsets where text will be drawn within the video frame. They are relative to the top/left border of the output image. The default value of x and y is "0". See below for the list of accepted constants and functions. The parameters for x and y are expressions containing the following constants and functions: dar input display aspect ratio, it is the same as (w / h) * sar hsub, vsub horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1. line_h, lh the height of each text line main_h, h, H the input height main_w, w, W the input width max_glyph_a, ascent the maximum distance from the baseline to the highest/upper grid coordinate used to place a glyph outline point, for all the rendered glyphs. It is a positive value, due to the grid's orientation with the Y axis upwards. max_glyph_d, descent the maximum distance from the baseline to the lowest grid coordinate used to place a glyph outline point, for all the rendered glyphs. This is a negative value, due to the grid's orientation, with the Y axis upwards. max_glyph_h maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent. max_glyph_w maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text n the number of input frame, starting from 0 rand(min, max) return a random number included between min and max sar input sample aspect ratio t timestamp expressed in seconds, NAN if the input timestamp is unknown text_h, th the height of the rendered text text_w, tw the width of the rendered text x, y the x and y offset coordinates where the text is drawn. These parameters allow the x and y expressions to refer each other, so you can for example specify "y=x/dar". If libavfilter was built with "--enable-fontconfig", then fontfile can be a fontconfig pattern or omitted. Some examples follow. · Draw "Test Text" with font FreeSerif, using the default values for the optional parameters. drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" · Draw 'Test Text' with font FreeSerif of size 24 at position x=100 and y=50 (counting from the top-left corner of the screen), text is yellow with a red box around it. Both the text and the box have an opacity of 20%. drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\ x=100: y=50: fontsize=24<EMAIL_ADDRESS>box=1<EMAIL_ADDRESS>Note that the double quotes are not necessary if spaces are not used within the parameter list. · Show the text at the center of the video frame: drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2" · Show a text line sliding from right to left in the last row of the video frame. The file LONG_LINE is assumed to contain a single line with no newlines. drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" · Show the content of file CREDITS off the bottom of the frame and scroll up. drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t" · Draw a single green letter "g", at the center of the input video. The glyph baseline is placed at half screen height. drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent" · Show text for 1 second every 3 seconds: drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:draw=lt(mod(t\\,3)\\,1):text='blink'" · Use fontconfig to set the font. Note that the colons need to be escaped. drawtext='fontfile=Linux Libertine O-40\\:style=Semibold:text=FFmpeg' For more information about libfreetype, check: <http://www.freetype.org/>. For more information about fontconfig, check: <http://freedesktop.org/software/fontconfig/fontconfig-user.html>. edgedetect Detect and draw edges. The filter uses the Canny Edge Detection algorithm. This filter accepts the following optional named parameters: low, high Set low and high threshold values used by the Canny thresholding algorithm. The high threshold selects the "strong" edge pixels, which are then connected through 8-connectivity with the "weak" edge pixels selected by the low threshold. low and high threshold values must be choosen in the range [0,1], and low should be lesser or equal to high. Default value for low is "20/255", and default value for high is "50/255". Example: edgedetect=low=0.1:high=0.4 fade Apply fade-in/out effect to input video. It accepts the parameters: type:start_frame:nb_frames[:options] type specifies if the effect type, can be either "in" for fade-in, or "out" for a fade-out effect. start_frame specifies the number of the start frame for starting to apply the fade effect. nb_frames specifies the number of frames for which the fade effect has to last. At the end of the fade-in effect the output video will have the same intensity as the input video, at the end of the fade-out transition the output video will be completely black. options is an optional sequence of key=value pairs, separated by ":". The description of the accepted options follows. type, t See type. start_frame, s See start_frame. nb_frames, n See nb_frames. alpha If set to 1, fade only alpha channel, if one exists on the input. Default value is 0. A few usage examples follow, usable too as test scenarios. # fade in first 30 frames of video fade=in:0:30 # fade out last 45 frames of a 200-frame video fade=out:155:45 # fade in first 25 frames and fade out last 25 frames of a 1000-frame video fade=in:0:25, fade=out:975:25 # make first 5 frames black, then fade in from frame 5-24 fade=in:5:20 # fade in alpha over first 25 frames of video fade=in:0:25:alpha=1 fieldorder Transform the field order of the input video. It accepts one parameter which specifies the required field order that the input interlaced video will be transformed to. The parameter can assume one of the following values: 0 or bff output bottom field first 1 or tff output top field first Default value is "tff". Transformation is achieved by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. This method is consistent with most broadcast field order converters. If the input video is not flagged as being interlaced, or it is already flagged as being of the required output field order then this filter does not alter the incoming video. This filter is very useful when converting to or from PAL DV material, which is bottom field first. For example: ffmpeg -i in.vob -vf "fieldorder=bff" out.dv fifo Buffer input images and send them when they are requested. This filter is mainly useful when auto-inserted by the libavfilter framework. The filter does not take parameters. format Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is supported for the input to the next filter. The filter accepts a list of pixel format names, separated by ":", for example "yuv420p:monow:rgb24". Some examples follow: # convert the input video to the format "yuv420p" format=yuv420p # convert the input video to any of the formats in the list format=yuv420p:yuv444p:yuv410p fps Convert the video to specified constant framerate by duplicating or dropping frames as necessary. This filter accepts the following named parameters: fps Desired output framerate. framestep Select one frame every N. This filter accepts in input a string representing a positive integer. Default argument is 1. frei0r Apply a frei0r effect to the input video. To enable compilation of this filter you need to install the frei0r header and configure FFmpeg with "--enable-frei0r". The filter supports the syntax: <filter_name>[{:|=}<param1>:<param2>:...:<paramN>] filter_name is the name to the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r effect is searched in each one of the directories specified by the colon separated list in FREIOR_PATH, otherwise in the standard frei0r paths, which are in this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/, /usr/lib/frei0r-1/. param1, param2, ... , paramN specify the parameters for the frei0r effect. A frei0r effect parameter can be a boolean (whose values are specified with "y" and "n"), a double, a color (specified by the syntax R/G/B, R, G, and B being float numbers from 0.0 to 1.0) or by an "av_parse_color()" color description), a position (specified by the syntax X/Y, X and Y being float numbers) and a string. The number and kind of parameters depend on the loaded effect. If an effect parameter is not specified the default value is set. Some examples follow: · Apply the distort0r effect, set the first two double parameters: frei0r=distort0r:0.5:0.01 · Apply the colordistance effect, takes a color as first parameter: frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 · Apply the perspective effect, specify the top left and top right image positions: frei0r=perspective:0.2/0.2:0.8/0.2 For more information see: <http://frei0r.dyne.org> gradfun Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8bit color depth. Interpolate the gradients that should go where the bands are, and dither them. This filter is designed for playback only. Do not use it prior to lossy compression, because compression tends to lose the dither and bring back the bands. The filter takes two optional parameters, separated by ':': strength:radius strength is the maximum amount by which the filter will change any one pixel. Also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 255, default value is 1.2, out-of-range values will be clipped to the valid range. radius is the neighborhood to fit the gradient to. A larger radius makes for smoother gradients, but also prevents the filter from modifying the pixels near detailed regions. Acceptable values are 8-32, default value is 16, out-of-range values will be clipped to the valid range. # default parameters gradfun=1.2:16 # omitting radius gradfun=1.2 hflip Flip the input video horizontally. For example to horizontally flip the input video with ffmpeg: ffmpeg -i in.avi -vf "hflip" out.avi hqdn3d High precision/quality 3d denoise filter.  It should enhance compressibility. It accepts the following optional parameters: luma_spatial:chroma_spatial:luma_tmp:chroma_tmp luma_spatial a non-negative float number which specifies spatial luma strength, defaults to 4.0 chroma_spatial a non-negative float number which specifies spatial chroma strength, defaults to 3.0*luma_spatial/4.0 luma_tmp a float number which specifies luma temporal strength, defaults to 6.0*luma_spatial/4.0 chroma_tmp a float number which specifies chroma temporal strength, defaults to luma_tmp*chroma_spatial/luma_spatial hue Modify the hue and/or the saturation of the input. This filter accepts the following optional named options: h Specify the hue angle as a number of degrees. It accepts a float number or an expression, and defaults to 0.0. H Specify the hue angle as a number of degrees. It accepts a float number or an expression, and defaults to 0.0. s Specify the saturation in the [-10,10] range. It accepts a float number and defaults to 1.0. The h, H and s parameters are expressions containing the following constants: n frame count of the input frame starting from 0 pts presentation timestamp of the input frame expressed in time base units r frame rate of the input video, NAN if the input frame rate is unknown t timestamp expressed in seconds, NAN if the input timestamp is unknown tb time base of the input video The options can also be set using the syntax: hue:saturation In this case hue is expressed in degrees. Some examples follow: · Set the hue to 90 degrees and the saturation to 1.0: hue=h=90:s=1 · Same command but expressing the hue in radians: hue=H=PI/2:s=1 · Same command without named options, hue must be expressed in degrees: hue=90:1 · Note that "h:s" syntax does not support expressions for the values of h and s, so the following example will issue an error: hue=PI/2:1 · Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second: hue="H=2*PI*t: s=sin(2*PI*t)+1" · Apply a 3 seconds saturation fade-in effect starting at 0: hue="s=min(t/3\,1)" The general fade-in expression can be written as: hue="s=min(0\, max((t-START)/DURATION\, 1))" · Apply a 3 seconds saturation fade-out effect starting at 5 seconds: hue="s=max(0\, min(1\, (8-t)/3))" The general fade-out expression can be written as: hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))" Commands This filter supports the following command: reinit Modify the hue and/or the saturation of the input video. The command accepts the same named options and syntax than when calling the filter from the command-line. If a parameter is omitted, it is kept at its current value. idet Interlaceing detect filter. This filter tries to detect if the input is interlaced or progressive. Top or bottom field first. lut, lutrgb, lutyuv Compute a look-up table for binding each pixel component input value to an output value, and apply it to input video. lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video. These filters accept in input a ":"-separated list of options, which specify the expressions used for computing the lookup table for the corresponding pixel component values. The lut filter requires either YUV or RGB pixel formats in input, and accepts the options: c0 first pixel component c1 second pixel component c2 third pixel component c3 fourth pixel component, corresponds to the alpha component The exact component associated to each option depends on the format in input. The lutrgb filter requires RGB pixel formats in input, and accepts the options: r red component g green component b blue component a alpha component The lutyuv filter requires YUV pixel formats in input, and accepts the options: y Y/luminance component u U/Cb component v V/Cr component a alpha component The expressions can contain the following constants and functions: w, h the input width and height val input value for the pixel component clipval the input value clipped in the minval-maxval range maxval maximum value for the pixel component minval minimum value for the pixel component negval the negated value for the pixel component value clipped in the minval-maxval range , it corresponds to the expression "maxval-clipval+minval" clip(val) the computed value in val clipped in the minval-maxval range gammaval(gamma) the computed gamma correction value of the pixel component value clipped in the minval-maxval range, corresponds to the expression "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval" All expressions default to "val". Some examples follow: # negate input video lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val" lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val" # the above is the same as lutrgb="r=negval:g=negval:b=negval" lutyuv="y=negval:u=negval:v=negval" # negate luminance lutyuv=y=negval # remove chroma components, turns the video into a graytone image lutyuv="u=128:v=128" # apply a luma burning effect lutyuv="y=2*val" # remove green and blue components lutrgb="g=0:b=0" # set a constant alpha channel value on input format=rgba,lutrgb=a="maxval-minval/2" # correct luminance gamma by a 0.5 factor lutyuv=y=gammaval(0.5) mp Apply an MPlayer filter to the input video. This filter provides a wrapper around most of the filters of MPlayer/MEncoder. This wrapper is considered experimental. Some of the wrapped filters may not work properly and we may drop support for them, as they will be implemented natively into FFmpeg. Thus you should avoid depending on them when writing portable scripts. The filters accepts the parameters: filter_name[:=]filter_params filter_name is the name of a supported MPlayer filter, filter_params is a string containing the parameters accepted by the named filter. The list of the currently supported filters follows: denoise3d detc dint divtc down3dright dsize eq2 eq field fil fixpts fspp geq harddup hqdn3d il ilpack ivtc kerndeint mcdeint noise ow palette perspective phase pp7 pullup qp rectangle sab softpulldown softskip spp telecine tile tinterlace unsharp uspp yuvcsp yvu9 The parameter syntax and behavior for the listed filters are the same of the corresponding MPlayer filters. For detailed instructions check the "VIDEO FILTERS" section in the MPlayer manual. Some examples follow: · Adjust gamma, brightness, contrast: mp=eq2=1.0:2:0.5 · Add temporal noise to input video: mp=noise=20t See also mplayer(1), <http://www.mplayerhq.hu/>. negate Negate input video. This filter accepts an integer in input, if non-zero it negates the alpha component (if available). The default value in input is 0. noformat Force libavfilter not to use any of the specified pixel formats for the input to the next filter. The filter accepts a list of pixel format names, separated by ":", for example "yuv420p:monow:rgb24". Some examples follow: # force libavfilter to use a format different from "yuv420p" for the # input to the vflip filter noformat=yuv420p,vflip # convert the input video to any of the formats not contained in the list noformat=yuv420p:yuv444p:yuv410p null Pass the video source unchanged to the output. ocv Apply video transform using libopencv. To enable this filter install libopencv library and headers and configure FFmpeg with "--enable-libopencv". The filter takes the parameters: filter_name{:=}filter_params. filter_name is the name of the libopencv filter to apply. filter_params specifies the parameters to pass to the libopencv filter. If not specified the default values are assumed. Refer to the official libopencv documentation for more precise information: <http://opencv.willowgarage.com/documentation/c/image_filtering.html> Follows the list of supported libopencv filters. dilate Dilate an image by using a specific structuring element. This filter corresponds to the libopencv function "cvDilate". It accepts the parameters: struct_el:nb_iterations. struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y the anchor point, and shape the shape for the structuring element, and can be one of the values "rect", "cross", "ellipse", "custom". If the value for shape is "custom", it must be followed by a string of the form "=filename". The file with name filename is assumed to represent a binary image, with each printable character corresponding to a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of the read file are assumed instead. The default value for struct_el is "3x3+0x0/rect". nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1. Follow some example: # use the default values ocv=dilate # dilate using a structuring element with a 5x5 cross, iterate two times ocv=dilate=5x5+2x2/cross:2 # read the shape from the file diamond.shape, iterate two times # the file diamond.shape may contain a pattern of characters like this: # * # *** # ***** # *** # * # the specified cols and rows are ignored (but not the anchor point coordinates) ocv=0x0+2x2/custom=diamond.shape:2 erode Erode an image by using a specific structuring element. This filter corresponds to the libopencv function "cvErode". The filter accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter. smooth Smooth the input video. The filter takes the following parameters: type:param1:param2:param3:param4. type is the type of smooth filter to apply, and can be one of the following values: "blur", "blur_no_scale", "median", "gaussian", "bilateral". The default value is "gaussian". param1, param2, param3, and param4 are parameters whose meanings depend on smooth type. param1 and param2 accept integer positive values or 0, param3 and param4 accept float values. The default value for param1 is 3, the default value for the other parameters is 0. These parameters correspond to the parameters assigned to the libopencv function "cvSmooth". overlay Overlay one video on top of another. It takes two inputs and one output, the first input is the "main" video on which the second input is overlayed. It accepts the parameters: x:y[:options]. x is the x coordinate of the overlayed video on the main video, y is the y coordinate. x and y are expressions containing the following parameters: main_w, main_h main input width and height W, H same as main_w and main_h overlay_w, overlay_h overlay input width and height w, h same as overlay_w and overlay_h options is an optional list of key=value pairs, separated by ":". The description of the accepted options follows. rgb If set to 1, force the filter to accept inputs in the RGB color space. Default value is 0. Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as it does the example for the movie filter. Follow some examples: # draw the overlay at 10 pixels from the bottom right # corner of the main video. overlay=main_w-overlay_w-10:main_h-overlay_h-10 # insert a transparent PNG logo in the bottom left corner of the input ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output # insert 2 different transparent PNG logos (second logo on bottom # right corner): ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=10:H-h-10,overlay=W-w-10:H-h-10' output # add a transparent color layer on top of the main video, # WxH specifies the size of the main input to the overlay filter color=red.3:WxH [over]; [in][over] overlay [out] # play an original video and a filtered version (here with the deshake filter) # side by side ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w' # the previous example is the same as: ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w' You can chain together more overlays but the efficiency of such approach is yet to be tested. pad Add paddings to the input image, and places the original input at the given coordinates x, y. It accepts the following parameters: width:height:x:y:color. The parameters width, height, x, and y are expressions containing the following constants: in_w, in_h the input video width and height iw, ih same as in_w and in_h out_w, out_h the output width and height, that is the size of the padded area as specified by the width and height expressions ow, oh same as out_w and out_h x, y x and y offsets as specified by the x and y expressions, or NAN if not yet specified a same as iw / ih sar input sample aspect ratio dar input display aspect ratio, it is the same as (iw / ih) * sar hsub, vsub horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1. Follows the description of the accepted parameters. width, height Specify the size of the output image with the paddings added. If the value for width or height is 0, the corresponding input size is used for the output. The width expression can reference the value set by the height expression, and vice versa. The default value of width and height is 0. x, y Specify the offsets where to place the input image in the padded area with respect to the top/left border of the output image. The x expression can reference the value set by the y expression, and vice versa. The default value of x and y is 0. color Specify the color of the padded area, it can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence. The default value of color is "black". Examples · Add paddings with color "violet" to the input video. Output video size is 640x480, the top-left corner of the input video is placed at column 0, row 40: pad=640:480:0:40:violet · Pad the input to get an output with dimensions increased by 3/2, and put the input video at the center of the padded area: pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" · Pad the input to get a squared output with size equal to the maximum value between the input width and height, and put the input video at the center of the padded area: pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" · Pad the input to get a final w/h ratio of 16:9: pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" · In case of anamorphic video, in order to set the output display aspect correctly, it is necessary to use sar in the expression, according to the relation: (ih * X / ih) * sar = output_dar X = output_dar / sar Thus the previous example needs to be modified to: pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" · Double output size and put the input video in the bottom-right corner of the output padded area: pad="2*iw:2*ih:ow-iw:oh-ih" pixdesctest Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video. For example: format=monow, pixdesctest can be used to test the monowhite pixel format descriptor definition. removelogo Suppress a TV station logo, using an image file to determine which pixels comprise the logo. It works by filling in the pixels that comprise the logo with neighboring pixels. This filter requires one argument which specifies the filter bitmap file, which can be any image format supported by libavformat. The width and height of the image file must match those of the video stream being processed. Pixels in the provided bitmap image with a value of zero are not considered part of the logo, non-zero pixels are considered part of the logo. If you use white (255) for the logo and black (0) for the rest, you will be safe. For making the filter bitmap, it is recommended to take a screen capture of a black frame with the logo visible, and then using a threshold filter followed by the erode filter once or twice. If needed, little splotches can be fixed manually. Remember that if logo pixels are not covered, the filter quality will be much reduced. Marking too many pixels as part of the logo does not hurt as much, but it will increase the amount of blurring needed to cover over the image and will destroy more information than necessary, and extra pixels will slow things down on a large logo. scale Scale the input video to width:height[:interl={1|-1}] and/or convert the image format. The scale filter forces the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio. The parameters width and height are expressions containing the following constants: in_w, in_h the input width and height iw, ih same as in_w and in_h out_w, out_h the output (cropped) width and height ow, oh same as out_w and out_h a same as iw / ih sar input sample aspect ratio dar input display aspect ratio, it is the same as (iw / ih) * sar hsub, vsub horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1. 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. Valid values for the optional parameter interl are: 1 force interlaced aware scaling -1 select interlaced aware scaling depending on whether the source frames are flagged as interlaced or not Unless interl is set to one of the above options, interlaced scaling will not be used. Some examples follow: # scale the input video to a size of 200x100. scale=200:100 # scale the input to 2x scale=2*iw:2*ih # the above is the same as scale=2*in_w:2*in_h # scale the input to 2x with forced interlaced scaling scale=2*iw:2*ih:interl=1 # scale the input to half size scale=iw/2:ih/2 # increase the width, and set the height to the same size scale=3/2*iw:ow # seek for Greek harmony scale=iw:1/PHI*iw scale=ih*PHI:ih # increase the height, and set the width to 3/2 of the height scale=3/2*oh:3/5*ih # increase the size, but make the size a multiple of the chroma scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" # increase the width to a maximum of 500 pixels, keep the same input aspect ratio scale='min(500\, iw*3/2):-1' select Select frames to pass in output. It accepts in input an expression, which is evaluated for each input frame. If the expression is evaluated to a non-zero value, the frame is selected and passed to the output, otherwise it is discarded. The expression can contain the following constants: n the sequential number of the filtered frame, starting from 0 selected_n the sequential number of the selected frame, starting from 0 prev_selected_n the sequential number of the last selected frame, NAN if undefined TB timebase of the input timestamps pts the PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units, NAN if undefined t the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined prev_pts the PTS of the previously filtered video frame, NAN if undefined prev_selected_pts the PTS of the last previously filtered video frame, NAN if undefined prev_selected_t the PTS of the last previously selected video frame, NAN if undefined start_pts the PTS of the first video frame in the video, NAN if undefined start_t the time of the first video frame in the video, NAN if undefined pict_type the type of the filtered frame, can assume one of the following values: I P B S SI SP BI interlace_type the frame interlace type, can assume one of the following values: PROGRESSIVE the frame is progressive (not interlaced) TOPFIRST the frame is top-field-first BOTTOMFIRST the frame is bottom-field-first key 1 if the filtered frame is a key-frame, 0 otherwise pos the position in the file of the filtered frame, -1 if the information is not available (e.g. for synthetic video) scene value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one (see the example below) The default value of the select expression is "1". Some examples follow: # select all frames in input select # the above is the same as: select=1 # skip all frames: select=0 # select only I-frames select='eq(pict_type\,I)' # select one frame every 100 select='not(mod(n\,100))' # select only frames contained in the 10-20 time interval select='gte(t\,10)*lte(t\,20)' # select only I frames contained in the 10-20 time interval select='gte(t\,10)*lte(t\,20)*eq(pict_type\,I)' # select frames with a minimum distance of 10 seconds select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' Complete example to create a mosaic of the first scenes: ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png Comparing scene against a value between 0.3 and 0.5 is generally a sane choice. setdar, setsar The "setdar" filter sets the Display Aspect Ratio for the filter output video. This is done by changing the specified Sample (aka Pixel) Aspect Ratio, according to the following equation: <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR> Keep in mind that the "setdar" filter does not modify the pixel dimensions of the video frame. Also the display aspect ratio set by this filter may be changed by later filters in the filterchain, e.g. in case of scaling or if another "setdar" or a "setsar" filter is applied. The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the filter output video. Note that as a consequence of the application of this filter, the output display aspect ratio will change according to the equation above. Keep in mind that the sample aspect ratio set by the "setsar" filter may be changed by later filters in the filterchain, e.g. if another "setsar" or a "setdar" filter is applied. The "setdar" and "setsar" filters accept a parameter string which represents the wanted aspect ratio. The parameter can be a floating point number string, an expression, or a string of the form num:den, where num and den are the numerator and denominator of the aspect ratio. If the parameter is not specified, it is assumed the value "0:1". For example to change the display aspect ratio to 16:9, specify: setdar=16:9 The example above is equivalent to: setdar=1.77777 To change the sample aspect ratio to 10:11, specify: setsar=10:11 setfield Force field for the output video frame. The "setfield" filter marks the interlace type field for the output frames. It does not change the input frame, but only sets the corresponding property, which affects how the frame is treated by following filters (e.g. "fieldorder" or "yadif"). It accepts a string parameter, which can assume the following values: auto Keep the same field property. bff Mark the frame as bottom-field-first. tff Mark the frame as top-field-first. prog Mark the frame as progressive. showinfo Show a line containing various information for each input video frame. The input video is not modified. The shown line contains a sequence of key/value pairs of the form key:value. A description of each shown parameter follows: n sequential number of the input frame, starting from 0 pts Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad. pts_time Presentation TimeStamp of the input frame, expressed as a number of seconds pos position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic video) fmt pixel format name sar sample aspect ratio of the input frame, expressed in the form num/den s size of the input frame, expressed in the form widthxheight i interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first) iskey 1 if the frame is a key frame, 0 otherwise type picture type of the input frame ("I" for an I-frame, "P" for a P-frame, "B" for a B-frame, "?" for unknown type). Check also the documentation of the "AVPictureType" enum and of the "av_get_picture_type_char" function defined in libavutil/avutil.h. checksum Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame plane_checksum Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]" slicify Pass the images of input video on to next video filter as multiple slices. ffmpeg -i in.avi -vf "s