neulab/tldr huggingface dataset:
Xvid's internal postprocessing filters: deblock-chroma (also see -vf pp) chroma deblock filter deblock-luma (also see -vf pp) luma deblock filter dering-luma (also see -vf pp) luma deringing filter dering-chroma (also see -vf pp) chroma deringing filter filmeffect (also see -vf noise) Adds artificial film grain to the video.  May increase perceived quality, while lowering true quality.
rendering methods: dr2 Activate direct rendering method 2. nodr2 Deactivate direct rendering method 2.
-xy <value> (also see -zoom) value<=8 Scale image by factor <value>. value>8 Set width to value and calculate height to keep correct aspect ratio.
-y <y> (also see -zoom) (MPlayer only) Scale image to height <y> (if software/hardware scaling is available).  Disables aspect calculations.
-zoom Allow software scaling, where available.  This will allow scaling with output drivers (like x11, fbdev) that do not support hardware scaling where MPlayer disables scaling by default for performance reasons.
Audio filters allow you to modify the audio stream and its properties. The syntax is:
-af <filter1[=parameter1:parameter2:...],filter2,...> Setup a chain of audio filters.
NOTE: To get a full list of available audio filters, see -af help.
Audio filters are managed in lists.  There are a few commands to manage the filter list.
-af-add <filter1[,filter2,...]> Appends the filters given as arguments to the filter list.
-af-pre <filter1[,filter2,...]> Prepends the filters given as arguments to the filter list.
-af-del <index1[,index2,...]> Deletes the filters at the given indexes.  Index numbers start at 0, negative numbers address the end of the list (-1 is the last).
-af-clr Completely empties the filter list.
Available filters are:
resample[=srate[:sloppy[:type]]] Changes the sample rate of the audio stream.  Can be used if you have a fixed frequency sound card or if you are stuck with an old sound card that is only capable of max 44.1kHz.  This filter is automatically enabled if necessary.  It only supports 16-bit integer and float in native-endian format as input. NOTE: With MEncoder, you need to also use -srate <srate>. <srate> output sample frequency in Hz.  The valid range for this parameter is 8000 to 192000.  If the input and output sample frequency are the same or if this parameter is omitted the filter is automatically unloaded.  A high sample frequency normally improves the audio quality, especially when used in combination with other filters. <sloppy> Allow (1) or disallow (0) the output frequency to differ slightly from the frequency given by <srate> (default: 1). Can be used if the startup of the playback is extremely slow. <type> Select which resampling method to use. 0: linear interpolation (fast, poor quality especially when upsampling) 1: polyphase filterbank and integer processing 2: polyphase filterbank and floating point processing (slow, best quality)
EXAMPLE: mplayer -af resample=44100:0:0 would set the output frequency of the resample filter to 44100Hz using exact output frequency scaling and linear interpolation.
lavcresample[=srate[:length[:linear[:count[:cutoff]]]]] Changes the sample rate of the audio stream to an integer <srate> in Hz.  It only supports the 16-bit native-endian format. NOTE: With MEncoder, you need to also use -srate <srate>. <srate> the output sample rate <length> length of the filter with respect to the lower sampling rate (default: 16) <linear> if 1 then filters will be linearly interpolated between polyphase entries <count> log2 of the number of polyphase entries (..., 10->1024, 11->2048, 12->4096, ...)  (default: 10->1024) <cutoff> cutoff frequency (0.0-1.0), default set depending upon filter length
lavcac3enc[=tospdif[:bitrate[:minchn]]] Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports 16-bit native-endian input format, maximum 6 channels. The output is big-endian when outputting a raw AC-3 stream, native-endian when outputting to S/PDIF.  The output sample rate of this filter is same with the input sample rate.  When input sample rate is 48kHz, 44.1kHz, or 32kHz, this filter directly use it.  Otherwise a resampling filter is auto-inserted before this filter to make the input and output sample rate be 48kHz.  You need to specify '-channels N' to make the decoder decode audio into N-channel, then the filter can encode the N-channel input to AC-3. <tospdif> Output raw AC-3 stream if zero or not set, output to S/PDIF for passthrough when <tospdif> is set non-zero. <bitrate> The bitrate to encode the AC-3 stream.  Set it to either 384 or 384000 to get 384kbits.  Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640 Default bitrate is based on the input channel number: 1ch: 96,  2ch: 192, 3ch: 224,  4ch: 384,  5ch: 448,  6ch: 448 <minchn> If the input channel number is less than <minchn>, the filter will detach itself (default: 5).
sweep[=speed] Produces a sine sweep. <0.0-1.0> Sine function delta, use very low values to hear the sweep.
sinesuppress[=freq:decay] Remove a sine at the specified frequency.  Useful to get rid of the 50/60Hz noise on low quality audio equipment.  It probably only works on mono input. <freq> The frequency of the sine which should be removed (in Hz) (default: 50) <decay> Controls the adaptivity (a larger value will make the filter adapt to amplitude and phase changes quicker, a smaller value will make the adaptation slower) (default: 0.0001).  Reasonable values are around 0.001.
bs2b[=option1:option2:...] Bauer stereophonic to binaural transformation using libbs2b. Improves the headphone listening experience by making the sound similar to that from loudspeakers, allowing each ear to hear both channels and taking into account the distance difference and the head shadowing effect.  It is applicable only to 2 channel audio. fcut=<300-1000> Set cut frequency in Hz. feed=<10-150> Set feed level for low frequencies in 0.1*dB. profile=<value> Several profiles are available for convenience: default will be used if nothing else was specified (fcut=700, feed=45) cmoy Chu Moy circuit implementation (fcut=700, feed=60) jmeier Jan Meier circuit implementation (fcut=650, feed=95)
If fcut or feed options are specified together with a profile, they will be applied on top of the selected profile.
hrtf[=flag] Head-related transfer function: Converts multichannel audio to 2 channel output for headphones, preserving the spatiality of the sound.
Flag  Meaning m     matrix decoding of the rear channel s     2-channel matrix decoding 0     no matrix decoding (default)
equalizer=[g1:g2:g3:...:g10] 10 octave band graphic equalizer, implemented using 10 IIR band pass filters.  This means that it works regardless of what type of audio is being played back.  The center frequencies for the 10 bands are:
No. frequency 0    31.25 Hz 1    62.50 Hz 2   125.00 Hz 3   250.00 Hz 4   500.00 Hz 5    1.00 kHz 6    2.00 kHz 7    4.00 kHz 8    8.00 kHz 9   16.00 kHz
If the sample rate of the sound being played is lower than the center frequency for a frequency band, then that band will be disabled.  A known bug with this filter is that the characteristics for the uppermost band are not completely symmetric if the sample rate is close to the center frequency of that band.  This problem can be worked around by upsampling the sound using the resample filter before it reaches this filter. <g1>:<g2>:<g3>:...:<g10> floating point numbers representing the gain in dB for each frequency band (-12-12)
EXAMPLE: mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi Would amplify the sound in the upper and lower frequency region while canceling it almost completely around 1kHz.
channels=nch[:nr:from1:to1:from2:to2:from3:to3:...] Can be used for adding, removing, routing and copying audio channels.  If only <nch> is given the default routing is used, it works as follows: If the number of output channels is bigger than the number of input channels empty channels are inserted (except mixing from mono to stereo, then the mono channel is repeated in both of the output channels).  If the number of output channels is smaller than the number of input channels the exceeding channels are truncated. <nch> number of output channels (1-8) <nr> number of routes (1-8) <from1:to1:from2:to2:from3:to3:...> Pairs of numbers between 0 and 7 that define where to route each channel.
EXAMPLE: mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi Would change the number of channels to 4 and set up 4 routes that swap channel 0 and channel 1 and leave channel 2 and 3 intact.  Observe that if media containing two channels was played back, channels 2 and 3 would contain silence but 0 and 1 would still be swapped. mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi Would change the number of channels to 6 and set up 4 routes that copy channel 0 to channels 0 to 3.  Channel 4 and 5 will contain silence.
format[=format] (also see -format) Convert between different sample formats.  Automatically enabled when needed by the sound card or another filter. <format> Sets the desired format.  The general form is 'sbe', where 's' denotes the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the number of bits per sample (16, 24 or 32) and 'e' denotes the endianness ('le' means little-endian, 'be' big-endian and 'ne' the endianness of the computer MPlayer is running on).  Valid values (amongst others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this rule that are also valid format specifiers: u8, s8, floatle, floatbe, floatne, mulaw, alaw, mpeg2, ac3 and imaadpcm.
volume[=v[:sc]] Implements software volume control.  Use this filter with caution since it can reduce the signal to noise ratio of the sound.  In most cases it is best to set the level for the PCM sound to max, leave this filter out and control the output level to your speakers with the master volume control of the mixer.  In case your sound card has a digital PCM mixer instead of an analog one, and you hear distortion, use the MASTER mixer instead.  If there is an external amplifier connected to the computer (this is almost always the case), the noise level can be minimized by adjusting the master level and the volume knob on the amplifier until the hissing noise in the background is gone. This filter has a second feature: It measures the overall maximum sound level and prints out that level when MPlayer exits.  This volume estimate can be used for setting the sound level in MEncoder such that the maximum dynamic range is utilized.  This feature currently only works with floating-point data, use e.g. -af-adv force=5, or use -af stats. NOTE: This filter is not reentrant and can therefore only be enabled once for every audio stream. <v> Sets the desired gain in dB for all channels in the stream from -200dB to +60dB, where -200dB mutes the sound completely and +60dB equals a gain of 1000 (default: 0). <sc> Turns soft clipping on (1) or off (0).  Soft-clipping can make the sound more smooth if very high volume levels are used.  Enable this option if the dynamic range of the loudspeakers is very low. WARNING: This feature creates distortion and should be considered a last resort.
EXAMPLE: mplayer -af volume=10.1:0 media.avi Would amplify the sound by 10.1dB and hard-clip if the sound level is too high.
pan=n[:L00:L01:L02:...L10:L11:L12:...Ln0:Ln1:Ln2:...] Mixes channels arbitrarily.  Basically a combination of the volume and the channels filter that can be used to down-mix many channels to only a few, e.g. stereo to mono or vary the "width" of the center speaker in a surround sound system.  This filter is hard to use, and will require some tinkering before the desired result is obtained.  The number of options for this filter depends on the number of output channels.  An example how to downmix a six- channel file to two channels with this filter can be found in the examples section near the end. <n> number of output channels (1-8) <Lij> How much of input channel i is mixed into output channel j (0-1).  So in principle you first have n numbers saying what to do with the first input channel, then n numbers that act on the second input channel etc.  If you do not specify any numbers for some input channels, 0 is assumed.
EXAMPLE: mplayer -af pan=1:0.5:0.5 media.avi Would down-mix from stereo to mono. mplayer -af pan=3:1:0:0.5:0:1:0.5 media.avi Would give 3 channel output leaving channels 0 and 1 intact, and mix channels 0 and 1 into output channel 2 (which could be sent to a subwoofer for example).
sub[=fc:ch] Adds a subwoofer channel to the audio stream.  The audio data used for creating the subwoofer channel is an average of the sound in channel 0 and channel 1.  The resulting sound is then low-pass filtered by a 4th order Butterworth filter with a default cutoff frequency of 60Hz and added to a separate channel in the audio stream. Warning: Disable this filter when you are playing DVDs with Dolby Digital 5.1 sound, otherwise this filter will disrupt the sound to the subwoofer. <fc> cutoff frequency in Hz for the low-pass filter (20Hz to 300Hz) (default: 60Hz) For the best result try setting the cutoff frequency as low as possible.  This will improve the stereo or surround sound experience. <ch> Determines the channel number in which to insert the sub- channel audio.  Channel number can be between 0 and 7 (default: 5).  Observe that the number of channels will automatically be increased to <ch> if necessary.
EXAMPLE: mplayer -af sub=100:4 -channels 5 media.avi Would add a sub-woofer channel with a cutoff frequency of 100Hz to output channel 4.
center Creates a center channel from the front channels.  May currently be low quality as it does not implement a high-pass filter for proper extraction yet, but averages and halves the channels instead. <ch> Determines the channel number in which to insert the center channel.  Channel number can be between 0 and 7 (default: 5).  Observe that the number of channels will automatically be increased to <ch> if necessary.
surround[=delay] Decoder for matrix encoded surround sound like Dolby Surround. Many files with 2 channel audio actually contain matrixed surround sound.  Requires a sound card supporting at least 4 channels. <delay> delay time in ms for the rear speakers (0 to 1000) (default: 20) This delay should be set as follows: If d1 is the distance from the listening position to the front speakers and d2 is the distance from the listening position to the rear speakers, then the delay should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
EXAMPLE: mplayer -af surround=15 -channels 4 media.avi Would add surround sound decoding with 15ms delay for the sound to the rear speakers.
delay[=ch1:ch2:...] Delays the sound to the loudspeakers such that the sound from the different channels arrives at the listening position simultaneously.  It is only useful if you have more than 2 loudspeakers. ch1,ch2,... The delay in ms that should be imposed on each channel (floating point number between 0 and 1000).
To calculate the required delay for the different channels do as follows:
1. Measure the distance to the loudspeakers in meters in relation to your listening position, giving you the distances s1 to s5 (for a 5.1 system).  There is no point in compensating for the subwoofer (you will not hear the difference anyway).
2. Subtract the distances s1 to s5 from the maximum distance, i.e. s[i] = max(s) - s[i]; i = 1...5.
3. Calculate the required delays in ms as d[i] = 1000*s[i]/342; i = 1...5.
EXAMPLE: mplayer -af delay=10.5:10.5:0:0:7:0 media.avi Would delay front left and right by 10.5ms, the two rear channels and the sub by 0ms and the center channel by 7ms.
export[=mmapped_file[:nsamples]] Exports the incoming signal to other processes using memory mapping (mmap()).  Memory mapped areas contain a header:
int nch                      /*number of channels*/ int size                     /*buffer size*/ unsigned long long counter   /*Used to keep sync, updated every time new data is exported.*/
The rest is payload (non-interleaved) 16 bit data. <mmapped_file> file to map data to (default: ~/.mplayer/mplayer- af_export) <nsamples> number of samples per channel (default: 512)
EXAMPLE: mplayer -af export=/tmp/mplayer-af_export:1024 media.avi Would export 1024 samples per channel to '/tmp/mplayer- af_export'.
extrastereo[=mul] (Linearly) increases the difference between left and right channels which adds some sort of "live" effect to playback. <mul> Sets the difference coefficient (default: 2.5).  0.0 means mono sound (average of both channels), with 1.0 sound will be unchanged, with -1.0 left and right channels will be swapped.
volnorm[=method:target] Maximizes the volume without distorting the sound. <method> Sets the used method. 1: Use a single sample to smooth the variations via the standard weighted mean over past samples (default). 2: Use several samples to smooth the variations via the standard weighted mean over past samples.
<target> Sets the target amplitude as a fraction of the maximum for the sample type (default: 0.25).
ladspa=file:label[:controls...] Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This filter is reentrant, so multiple LADSPA plugins can be used at once. <file> Specifies the LADSPA plugin library file.  If LADSPA_PATH is set, it searches for the specified file.  If it is not set, you must supply a fully specified pathname. <label> Specifies the filter within the library.  Some libraries contain only one filter, but others contain many of them. Entering 'help' here, will list all available filters within the specified library, which eliminates the use of 'listplugins' from the LADSPA SDK. <controls> Controls are zero or more floating point values that determine the behavior of the loaded plugin (for example delay, threshold or gain).  In verbose mode (add -v to the MPlayer command line), all available controls and their valid ranges are printed.  This eliminates the use of 'analyseplugin' from the LADSPA SDK.
comp Compressor/expander filter usable for microphone input.  Prevents artifacts on very loud sound and raises the volume on very low sound.  This filter is untested, maybe even unusable.
gate Noise gate filter similar to the comp audio filter.  This filter is untested, maybe even unusable.
karaoke Simple voice removal filter exploiting the fact that voice is usually recorded with mono gear and later 'center' mixed onto the final audio stream.  Beware that this filter will turn your signal into mono.  Works well for 2 channel tracks; do not bother trying it on anything but 2 channel stereo.
scaletempo[=option1:option2:...] Scales audio tempo without altering pitch, optionally synced to playback speed (default). This works by playing ´stride´ ms of audio at normal speed then consuming ´stride*scale´ ms of input audio.  It pieces the strides together by blending ´overlap´% of stride with audio following the previous stride.  It optionally performs a short statistical analysis on the next ´search´ ms of audio to determine the best overlap position. scale=<amount> Nominal amount to scale tempo.  Scales this amount in addition to speed.  (default: 1.0) stride=<amount> Length in milliseconds to output each stride.  Too high of value will cause noticeable skips at high scale amounts and an echo at low scale amounts.  Very low values will alter pitch.  Increasing improves performance.  (default: 60) overlap=<percent> Percentage of stride to overlap.  Decreasing improves performance.  (default: .20) search=<amount> Length in milliseconds to search for best overlap position.  Decreasing improves performance greatly.  On slow systems, you will probably want to set this very low. (default: 14) speed=<tempo|pitch|both|none> Set response to speed change. tempo Scale tempo in sync with speed (default). pitch Reverses effect of filter.  Scales pitch without altering tempo.  Add ´[ speed_mult 0.9438743126816935´ and ´] speed_mult 1.059463094352953´ to your input.conf to step by musical semi-tones.  WARNING: Loses sync with video. both Scale both tempo and pitch. none Ignore speed changes.
EXAMPLE: mplayer -af scaletempo -speed 1.2 media.ogg Would playback media at 1.2x normal speed, with audio at normal pitch.  Changing playback speed, would change audio tempo to match. mplayer -af scaletempo=scale=1.2:speed=none -speed 1.2 media.ogg Would playback media at 1.2x normal speed, with audio at normal pitch, but changing playback speed has no effect on audio tempo. mplayer -af scaletempo=stride=30:overlap=.50:search=10 media.ogg Would tweak the quality and performace parameters. mplayer -af format=floatne,scaletempo media.ogg Would make scaletempo use float code.  Maybe faster on some platforms. mplayer -af scaletempo=scale=1.2:speed=pitch audio.ogg Would playback audio file at 1.2x normal speed, with audio at normal pitch.  Changing playback speed, would change pitch, leaving audio tempo at 1.2x.
stats Collects and prints statistics about the audio stream, especially the volume.  These statistics are especially intended to help adjusting the volume while avoiding clipping.  The volumes are printed in dB and compatible with the volume audio filter, they are always rounded towards -0dB.
The 'n_samples' field is the total number of samples seen by the filter.  The 'mean_volume' field is the root mean square.  The 'max_volume' field is exactly what it says.  The 'histogram_Xdb' fields count how many samples were at -XdB, for X just below max_volume.
For example, if max_volume is -7dB and histogram_7dB is 19, 'volume=7' will not cause clipping and 'volume=8' will cause clipping on exactly 19 samples.
Video filters allow you to modify the video stream and its properties. The syntax is:
-vf <filter1[=parameter1:parameter2:...],filter2,...> Setup a chain of video filters.
Many parameters are optional and set to default values if omitted.  To explicitly use a default value set a parameter to '-1'.  Parameters w:h means width x height in pixels, x:y means x;y position counted from the upper left corner of the bigger image. NOTE: To get a full list of available video filters, see -vf help.
Video filters are managed in lists.  There are a few commands to manage the filter list.
-vf-add <filter1[,filter2,...]> Appends the filters given as arguments to the filter list.
-vf-pre <filter1[,filter2,...]> Prepends the filters given as arguments to the filter list.
-vf-del <index1[,index2,...]> Deletes the filters at the given indexes.  Index numbers start at 0, negative numbers address the end of the list (-1 is the last).
-vf-clr Completely empties the filter list.
With filters that support it, you can access parameters by their name.
-vf <filter>=help Prints the parameter names and parameter value ranges for a particular filter.
-vf <filter=named_parameter1=value1[:named_parameter2=value2:...]> Sets a named parameter to the given value.  Use on and off or yes and no to set flag parameters.
crop[=w:h:x:y] Crops the given part of the image and discards the rest.  Useful to remove black bands from widescreen movies. <w>,<h> Cropped width and height, defaults to original width and height. <x>,<y> Position of the cropped picture, defaults to center.
cropdetect[=limit:round[:reset]] Calculates necessary cropping parameters and prints the recommended parameters to stdout. <limit> Threshold, which can be optionally specified from nothing (0) to everything (255) (default: 24). <round> Value which the width/height should be divisible by (default: 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 (default: 0).  This can be useful when channel logos distort the video area.  0 indicates never reset and return the largest area encountered during playback.
rectangle[=w:h:x:y] Draws a rectangle of the requested width and height at the specified coordinates over the image and prints current rectangle parameters to the console.  This can be used to find optimal cropping parameters.  If you bind the input.conf directive 'change_rectangle' to keystrokes, you can move and resize the rectangle on the fly. <w>,<h> width and height (default: -1, maximum possible width where boundaries are still visible.) <x>,<y> top left corner position (default: -1, uppermost leftmost)
expand[=w:h:x:y:o:a:r] Expands (not scales) movie resolution to the given value and places the unscaled original at coordinates x, y.  Can be used for placing subtitles/OSD in the resulting black bands.
<w>,<h> Expanded width,height (default: original width,height). Negative values for w and h are treated as offsets to the original size.
EXAMPLE: expand=0:-50:0:0 Adds a 50 pixel border to the bottom of the picture.
<x>,<y> position of original image on the expanded image (default: center)
<o> OSD/subtitle rendering 0: disable (default) 1: enable
<a> Expands to fit an aspect instead of a resolution (default: 0).
EXAMPLE: expand=800:::::4/3 Expands to 800x600, unless the source is higher resolution, in which case it expands to fill a 4/3 aspect.
<r> Rounds up to make both width and height divisible by <r> (default: 1).
flip (also see -flip) Flips the image upside down.
mirror Mirrors the image on the Y axis.
rotate[=<0-7>] Rotates the image by 90 degrees and optionally flips it.  For values between 4-7 rotation is only done if the movie geometry is portrait and not landscape.
0    Rotate by 90 degrees clockwise and flip (default).
1    Rotate by 90 degrees clockwise.
2    Rotate by 90 degrees counterclockwise.
3    Rotate by 90 degrees counterclockwise and flip.
scale[=w:h[:interlaced[:chr_drop[:par[:par2[:presize[:noup[:arnd]]]]]]]] Scales the image with the software scaler (slow) and performs a YUV<->RGB colorspace conversion (also see -sws).
<w>,<h> scaled width/height (default: original width/height) NOTE: If -zoom is used, and underlying filters (including libvo) are incapable of scaling, it defaults to d_width/ d_height! 0:   scaled d_width/d_height -1:   original width/height -2:   Calculate w/h using the other dimension and the prescaled aspect ratio. -3:   Calculate w/h using the other dimension and the original aspect ratio. -(n+8): Like -n above, but rounding the dimension to the closest multiple of 16.
<interlaced> Toggle interlaced scaling. 0: off (default) 1: on
<chr_drop> chroma skipping 0: Use all available input lines for chroma. 1: Use only every 2. input line for chroma. 2: Use only every 4. input line for chroma. 3: Use only every 8. input line for chroma.
<par>[:<par2>] (also see -sws) Set some scaling parameters depending on the type of scaler selected with -sws. -sws 2 (bicubic):  B (blurring) and C (ringing) 0.00:0.60 default 0.00:0.75 VirtualDub's "precise bicubic" 0.00:0.50 Catmull-Rom spline 0.33:0.33 Mitchell-Netravali spline 1.00:0.00 cubic B-spline -sws 7 (gaussian): sharpness (0 (soft) - 100 (sharp)) -sws 9 (lanczos):  filter length (1-10)
<presize> Scale to preset sizes. qntsc:   352x240 (NTSC quarter screen) qpal:    352x288 (PAL quarter screen) ntsc:    720x480 (standard NTSC) pal:     720x576 (standard PAL) sntsc:   640x480 (square pixel NTSC) spal:    768x576 (square pixel PAL)
<noup> Disallow upscaling past the original dimensions. 0: Allow upscaling (default). 1: Disallow upscaling if one dimension exceeds its original value. 2: Disallow upscaling if both dimensions exceed their original values.
<arnd> Accurate rounding for the vertical scaler, which may be faster or slower than the default rounding. 0: Disable accurate rounding (default). 1: Enable accurate rounding.
dsize[=aspect|w:h:aspect-method:r] Changes the intended display size/aspect at an arbitrary point in the filter chain.  Aspect can be given as a fraction (4/3) or floating point number (1.33).  Alternatively, you may specify the exact display width and height desired.  Note that this filter does not do any scaling itself; it just affects what later scalers (software or hardware) will do when auto-scaling to correct aspect.
<w>,<h> New display width and height.  Can also be these special values: 0:   original display width and height -1:   original video width and height (default) -2:   Calculate w/h using the other dimension and the original display aspect ratio. -3:   Calculate w/h using the other dimension and the original video aspect ratio.
EXAMPLE: dsize=800:-2 Specifies a display resolution of 800x600 for a 4/3 aspect video, or 800x450 for a 16/9 aspect video. <aspect-method> Modifies width and height according to original aspect ratios. -1: Ignore original aspect ratio (default). 0: Keep display aspect ratio by using <w> and <h> as maximum resolution. 1: Keep display aspect ratio by using <w> and <h> as minimum resolution. 2: Keep video aspect ratio by using <w> and <h> as maximum resolution. 3: Keep video aspect ratio by using <w> and <h> as minimum resolution.
EXAMPLE: dsize=800:600:0 Specifies a display resolution of at most 800x600, or smaller, in order to keep aspect.
yvu9 Forces software YVU9 to YV12 colorspace conversion.  Deprecated in favor of the software scaler.
yuvcsp Clamps YUV color values to the CCIR 601 range without doing real conversion.
palette RGB/BGR 8 -> 15/16/24/32bpp colorspace conversion using palette.
format[=fourcc[:outfourcc]] Restricts the colorspace for the next filter without doing any conversion.  Use together with the scale filter for a real conversion. NOTE: For a list of available formats see format=fmt=help. <fourcc> format name like rgb15, bgr24, yv12, etc (default: yuy2) <outfourcc> Format name that should be substituted for the output.  If this is not 100% compatible with the <fourcc> value it will crash. Valid examples: format=rgb24:bgr24 format=yuyv:yuy2 Invalid examples (will crash): format=rgb24:yv12
noformat[=fourcc] Restricts the colorspace for the next filter without doing any conversion.  Unlike the format filter, this will allow any colorspace except the one you specify. NOTE: For a list of available formats see noformat=fmt=help. <fourcc> format name like rgb15, bgr24, yv12, etc (default: yv12)
pp[=filter1[:option1[:option2...]]/[-]filter2...] (also see -pphelp) Enables the specified chain of postprocessing subfilters. Subfilters must be separated by '/' and can be disabled by prepending a '-'.  Each subfilter and some options have a short and a long name that can be used interchangeably, i.e. dr/dering are the same.  All subfilters share common options to determine their scope: a/autoq Automatically switch the subfilter off if the CPU is too slow. c/chrom Do chrominance filtering, too (default). y/nochrom Do luminance filtering only (no chrominance). n/noluma Do chrominance filtering only (no luminance).
NOTE: -pphelp shows a list of available subfilters.
Available subfilters are
hb/hdeblock[:difference[:flatness]] horizontal deblocking filter <difference>: Difference factor where higher values mean more deblocking (default: 32). <flatness>: Flatness threshold where lower values mean more deblocking (default: 39).
vb/vdeblock[:difference[:flatness]] vertical deblocking filter <difference>: Difference factor where higher values mean more deblocking (default: 32). <flatness>: Flatness threshold where lower values mean more deblocking (default: 39).
ha/hadeblock[:difference[:flatness]] accurate horizontal deblocking filter <difference>: Difference factor where higher values mean more deblocking (default: 32). <flatness>: Flatness threshold where lower values mean more deblocking (default: 39).
va/vadeblock[:difference[:flatness]] accurate vertical deblocking filter <difference>: Difference factor where higher values mean more deblocking (default: 32). <flatness>: Flatness threshold where lower values mean more deblocking (default: 39).
The horizontal and vertical deblocking filters share the difference and flatness values so you cannot set different horizontal and vertical thresholds.
h1/x1hdeblock experimental horizontal deblocking filter
v1/x1vdeblock experimental vertical deblocking filter
dr/dering deringing filter
tn/tmpnoise[:threshold1[:threshold2[:threshold3]]] temporal noise reducer <threshold1>: larger -> stronger filtering <threshold2>: larger -> stronger filtering <threshold3>: larger -> stronger filtering
al/autolevels[:f/fullyrange] automatic brightness / contrast correction f/fullyrange: Stretch luminance to (0-255).
lb/linblenddeint Linear blend deinterlacing filter that deinterlaces the given block by filtering all lines with a (1 2 1) filter.
li/linipoldeint Linear interpolating deinterlacing filter that deinterlaces the given block by linearly interpolating every second line.
ci/cubicipoldeint Cubic interpolating deinterlacing filter deinterlaces the given block by cubically interpolating every second line.
md/mediandeint Median deinterlacing filter that deinterlaces the given block by applying a median filter to every second line.
fd/ffmpegdeint FFmpeg deinterlacing filter that deinterlaces the given block by filtering every second line with a (-1 4 2 4 -1) filter.
l5/lowpass5 Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given block by filtering all lines with a (-1 2 6 2 -1) filter.
fq/forceQuant[:quantizer] Overrides the quantizer table from the input with the constant quantizer you specify. <quantizer>: quantizer to use
de/default default pp filter combination (hb:a,vb:a,dr:a)
fa/fast fast pp filter combination (h1:a,v1:a,dr:a)
ac high quality pp filter combination (ha:a:128:7,va:a,dr:a)
EXAMPLE: -vf pp=hb/vb/dr/al horizontal and vertical deblocking, deringing and automatic brightness/contrast -vf pp=de/-al default filters without brightness/contrast correction -vf pp=default/tmpnoise:1:2:3 Enable default filters & temporal denoiser. -vf pp=hb:y/vb:a Horizontal deblocking on luminance only, and switch vertical deblocking on or off automatically depending on available CPU time.
spp[=quality[:qp[:mode]]] Simple postprocessing filter that compresses and decompresses the image at several (or - in the case of quality level 6 - all) shifts and averages the results.
<quality> 0-6 (default: 3)
<qp> Force quantization parameter (default: 0, use QP from video).
<mode> 0: hard thresholding (default) 1: soft thresholding (better deringing, but blurrier) 4: like 0, but also use B-frames' QP (may cause flicker) 5: like 1, but also use B-frames' QP (may cause flicker)
uspp[=quality[:qp]] Ultra simple & slow postprocessing filter that compresses and decompresses the image at several (or - in the case of quality level 8 - all) shifts and averages the results.  The way this differs from the behavior of spp is that uspp actually encodes & decodes each case with libavcodec Snow, whereas spp uses a simplified intra only 8x8 DCT similar to MJPEG.
<quality> 0-8 (default: 3)
fspp[=quality[:qp[:strength[:bframes]]]] faster version of the simple postprocessing filter
<quality> 4-5 (equivalent to spp; default: 4)
<-15-32> Filter strength, lower values mean more details but also more artifacts, while higher values make the image smoother but also blurrier (default: 0 - PSNR optimal).
<bframes> 0: do not use QP from B-frames (default) 1: use QP from B-frames too (may cause flicker)
pp7[=qp[:mode]] Variant of the spp filter, similar to spp=6 with 7 point DCT where only the center sample is used after IDCT.
<mode> 0: hard thresholding 1: soft thresholding (better deringing, but blurrier) 2: medium thresholding (default, good results)
qp=equation quantization parameter (QP) change filter
<equation> some equation like "2+2*sin(PI*qp)"
geq=equation generic equation change filter
<equation> Some equation, e.g.  'p(W-X\,Y)' to flip the image horizontally.  You can use whitespace to make the equation more readable.  There are a couple of constants that can be used in the equation: PI: the number pi E: the number e X / Y: the coordinates of the current sample W / H: width and height of the image SW / SH: width/height scale depending on the currently filtered plane, e.g. 1,1 and 0.5,0.5 for YUV 4:2:0. p(x,y): returns the value of the pixel at location x/y of the current plane.
test Generate various test patterns.
rgbtest[=width:height] Generate an RGB test pattern useful for detecting RGB vs BGR issues.  You should see a red, green and blue stripe from top to bottom.
<width> Desired width of generated image (default: 0).  0 means width of input image.
<height> Desired height of generated image (default: 0).  0 means height of input image.
lavc[=quality:fps] Fast software YV12 to MPEG-1 conversion with libavcodec for use with DVB/DXR3/IVTV/V4L2.
<quality> 1-31: fixed qscale 32-:  fixed bitrate in kbits
<fps> force output fps (float value) (default: 0, autodetect based on height)
dvbscale[=aspect] Set up optimal scaling for DVB cards, scaling the x axis in hardware and calculating the y axis scaling in software to keep aspect.  Only useful together with expand and scale.
<aspect> Control aspect ratio, calculate as DVB_HEIGHT*ASPECTRATIO (default: 576*4/3=768), set it to 576*(16/9)=1024 for a 16:9 TV.
EXAMPLE: -vf dvbscale,scale=-1:0,expand=-1:576:-1:-1:1,lavc FIXME: Explain what this does.
noise[=luma[u][t|a][h][p]:chroma[u][t|a][h][p]] Adds noise. <0-100> luma noise <0-100> chroma noise u    uniform noise (gaussian otherwise) t    temporal noise (noise pattern changes between frames) a    averaged temporal noise (smoother, but a lot slower) h    high quality (slightly better looking, slightly slower) p    mix random noise with a (semi)regular pattern
denoise3d[=luma_spatial:chroma_spatial:luma_tmp:chroma_tmp] This filter aims to reduce image noise producing smooth images and making still images really still (This should enhance compressibility.). <luma_spatial> spatial luma strength (default: 4) <chroma_spatial> spatial chroma strength (default: 3) <luma_tmp> luma temporal strength (default: 6) <chroma_tmp> chroma temporal strength (default: luma_tmp*chroma_spatial/luma_spatial)
hqdn3d[=luma_spatial:chroma_spatial:luma_tmp:chroma_tmp] High precision/quality version of the denoise3d filter. Parameters and usage are the same.
ow[=depth[:luma_strength[:chroma_strength]]] Overcomplete Wavelet denoiser. <depth> Larger depth values will denoise lower frequency components more, but slow down filtering (default: 8). <luma_strength> luma strength (default: 1.0) <chroma_strength> chroma strength (default: 1.0)
eq[=brightness:contrast] (OBSOLETE) Software equalizer with interactive controls just like the hardware equalizer, for cards/drivers that do not support brightness and contrast controls in hardware.  Might also be useful with MEncoder, either for fixing poorly captured movies, or for slightly reducing contrast to mask artifacts and get by with lower bitrates. <-100-100> initial brightness <-100-100> initial contrast
eq2[=gamma:contrast:brightness:saturation:rg:gg:bg:weight] Alternative software equalizer that uses lookup tables (very slow), allowing gamma correction in addition to simple brightness and contrast adjustment.  Note that it uses the same MMX optimized code as -vf eq if all gamma values are 1.0.  The parameters are given as floating point values. <0.1-10> initial gamma value (default: 1.0) <-2-2> initial contrast, where negative values result in a negative image (default: 1.0) <-1-1> initial brightness (default: 0.0) <0-3> initial saturation (default: 1.0) <0.1-10> gamma value for the red component (default: 1.0) <0.1-10> gamma value for the green component (default: 1.0) <0.1-10> gamma value for the blue component (default: 1.0) <0-1> The weight parameter can be used to reduce the effect of a high gamma value on bright image areas, e.g. keep them from getting overamplified and just plain white.  A value of 0.0 turns the gamma correction all the way down while 1.0 leaves it at its full strength (default: 1.0).
hue[=hue:saturation] Software equalizer with interactive controls just like the hardware equalizer, for cards/drivers that do not support hue and saturation controls in hardware. <-180-180> initial hue (default: 0.0) <-100-100> initial saturation, where negative values result in a negative chroma (default: 1.0)
halfpack[=f] Convert planar YUV 4:2:0 to half-height packed 4:2:2, downsampling luma but keeping all chroma samples.  Useful for output to low- resolution display devices when hardware downscaling is poor quality or is not available.  Can also be used as a primitive luma-only deinterlacer with very low CPU usage. <f> By default, halfpack averages pairs of lines when downsampling.  Any value different from 0 or 1 gives the default (averaging) behavior. 0: Only use even lines when downsampling. 1: Only use odd lines when downsampling.
ilpack[=mode] When interlaced video is stored in YUV 4:2:0 formats, chroma interlacing does not line up properly due to vertical downsampling of the chroma channels.  This filter packs the planar 4:2:0 data into YUY2 (4:2:2) format with the chroma lines in their proper locations, so that in any given scanline, the luma and chroma data both come from the same field. <mode> Select the sampling mode. 0: nearest-neighbor sampling, fast but incorrect 1: linear interpolation (default)
harddup Only useful with MEncoder.  If harddup is used when encoding, it will force duplicate frames to be encoded in the output.  This uses slightly more space, but is necessary for output to MPEG files or if you plan to demux and remux the video stream after encoding.  Should be placed at or near the end of the filter chain unless you have a good reason to do otherwise.
softskip Only useful with MEncoder.  Softskip moves the frame skipping (dropping) step of encoding from before the filter chain to some point during the filter chain.  This allows filters which need to see all frames (inverse telecine, temporal denoising, etc.) to function properly.  Should be placed after the filters which need to see all frames and before any subsequent filters that are CPU- intensive.
decimate[=max:hi:lo:frac] 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. <max> Sets the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). <hi>,<lo>,<frac> A frame is a candidate for dropping if no 8x8 region differs by more than a threshold of <hi>, and if not more than <frac> portion (1 meaning the whole image) differs by more than a threshold of <lo>.  Values of <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.
dint[=sense:level] The drop-deinterlace (dint) filter detects and drops the first from a set of interlaced video frames. <0.0-1.0> relative difference between neighboring pixels (default: 0.1) <0.0-1.0> What part of the image has to be detected as interlaced to drop the frame (default: 0.15).
lavcdeint (OBSOLETE) FFmpeg deinterlacing filter, same as -vf pp=fd
lavfi=filtergraph FFmpeg libavfilter wrapper.  filtergraph defines a whole libavfilter graph with one input and one output.  See http://www.ffmpeg.org/libavfilter.html#SEC4 for details.
As a special case, if filtergraph is $word then the value of the word environment variable is used; this is necessary if commas are present in the graph description, as mplayer uses them as a delimiter between filters.
NOTE: This filter is considered experimental, it may interact strangely with other filters.
EXAMPLE: overlay="movie=$small_video, scale=160:120 [ca]; [in] [ca] overlay=16:8" mplayer -vf lavfi='$overlay' $big_video
kerndeint[=thresh[:map[:order[:sharp[:twoway]]]]] Donald Graft's adaptive kernel deinterlacer.  Deinterlaces parts of a video if a configurable threshold is exceeded. <0-255> threshold (default: 10) <map> 0: Ignore pixels exceeding the threshold (default). 1: Paint pixels exceeding the threshold white.
<order> 0: Leave fields alone (default). 1: Swap fields.
<sharp> 0: Disable additional sharpening (default). 1: Enable additional sharpening.
<twoway> 0: Disable twoway sharpening (default). 1: Enable twoway sharpening.
unsharp[=l|cWxH:amount[:l|cWxH:amount]] unsharp mask / gaussian blur
l Apply effect on luma component.
c Apply effect on chroma components.
<width>x<height> width and height of the matrix, odd sized in both directions (min = 3x3, max = 13x11 or 11x13, usually something between 3x3 and 7x7)
amount Relative amount of sharpness/blur to add to the image (a sane range should be -1.5-1.5). <0: blur >0: sharpen
swapuv Swap U & V plane.
il[=d|i][s][:[d|i][s]] (De)interleaves lines.  The goal of this filter is to add the ability to process interlaced images pre-field without deinterlacing them.  You can filter your interlaced DVD and play it on a TV without breaking the interlacing.  While deinterlacing (with the postprocessing filter) removes interlacing permanently (by smoothing, averaging, etc) deinterleaving splits the frame into 2 fields (so called half pictures), so you can process (filter) them independently and then re-interleave them. d    deinterleave (placing one above the other) i    interleave s    swap fields (exchange even & odd lines)
fil[=i|d] (De)interleaves lines.  This filter is very similar to the il filter but much faster, the main disadvantage is that it does not always work.  Especially if combined with other filters it may produce randomly messed up images, so be happy if it works but do not complain if it does not for your combination of filters. d    Deinterleave fields, placing them side by side. i    Interleave fields again (reversing the effect of fil=d).
field[=n] Extracts a single field from an interlaced image using stride arithmetic to avoid wasting CPU time.  The optional argument n specifies whether to extract the even or the odd field (depending on whether n is even or odd).
detc[=var1=value1:var2=value2:...] Attempts to reverse the 'telecine' process to recover a clean, non-interlaced stream at film framerate.  This was the first and most primitive inverse telecine filter to be added to MPlayer/ MEncoder.  It works by latching onto the telecine 3:2 pattern and following it as long as possible.  This makes it suitable for perfectly-telecined material, even in the presence of a fair degree of noise, but it will fail in the presence of complex post- telecine edits.  Development on this filter is no longer taking place, as ivtc, pullup, and filmdint are better for most applications.  The following arguments (see syntax above) may be used to control detc's behavior:
<dr> Set the frame dropping mode. 0: Do not drop frames to maintain fixed output framerate (default). 1: Always drop a frame when there have been no drops or telecine merges in the past 5 frames. 2: Always maintain exact 5:4 input to output frame ratio. NOTE: Use mode 1 or 2 with MEncoder.
<am> Analysis mode. 0: Fixed pattern with initial frame number specified by <fr>. 1: aggressive search for telecine pattern (default)
<fr> Set initial frame number in sequence.  0-2 are the three clean progressive frames; 3 and 4 are the two interlaced frames.  The default, -1, means 'not in telecine sequence'.  The number specified here is the type for the imaginary previous frame before the movie starts.
<t0>, <t1>, <t2>, <t3> Threshold values to be used in certain modes.
ivtc[=1] Experimental 'stateless' inverse telecine filter.  Rather than trying to lock on to a pattern like the detc filter does, ivtc makes its decisions independently for each frame.  This will give much better results for material that has undergone heavy editing after telecine was applied, but as a result it is not as forgiving of noisy input, for example TV capture.  The optional parameter (ivtc=1) corresponds to the dr=1 option for the detc filter, and should be used with MEncoder but not with MPlayer.  As with detc, you must specify the correct output framerate (-ofps 24000/1001) when using MEncoder.  Further development on ivtc has stopped, as the pullup and filmdint filters appear to be much more accurate.