{ "mode": "man", "parameter": "ffmpeg-filters", "section": "", "url": "https://www.chedong.com/phpMan.php/man/ffmpeg-filters/json", "generated": "2026-06-10T16:23:08Z", "sections": { "NAME": { "content": "ffmpeg-filters - FFmpeg filters\n", "subsections": [] }, "DESCRIPTION": { "content": "This document describes filters, sources, and sinks provided by the libavfilter library.\n", "subsections": [] }, "FILTERING INTRODUCTION": { "content": "Filtering in FFmpeg is enabled through the libavfilter library.\n\nIn libavfilter, a filter can have multiple inputs and multiple outputs. To illustrate the\nsorts of things that are possible, we consider the following filtergraph.\n\n[main]\ninput --> split ---------------------> overlay --> output\n| ^\n|[tmp] [flip]|\n+-----> crop --> vflip -------+\n\nThis filtergraph splits the input stream in two streams, then sends one stream through the\ncrop filter and the vflip filter, before merging it back with the other stream by overlaying\nit on top. You can use the following command to achieve this:\n\nffmpeg -i INPUT -vf \"split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2\" OUTPUT\n\nThe result will be that the top half of the video is mirrored onto the bottom half of the\noutput video.\n\nFilters in the same linear chain are separated by commas, and distinct linear chains of\nfilters are separated by semicolons. In our example, crop,vflip are in one linear chain,\nsplit and overlay are separately in another. The points where the linear chains join are\nlabelled by names enclosed in square brackets. In the example, the split filter generates two\noutputs that are associated to the labels [main] and [tmp].\n\nThe stream sent to the second output of split, labelled as [tmp], is processed through the\ncrop filter, which crops away the lower half part of the video, and then vertically flipped.\nThe overlay filter takes in input the first unchanged output of the split filter (which was\nlabelled as [main]), and overlay on its lower half the output generated by the crop,vflip\nfilterchain.\n\nSome filters take in input a list of parameters: they are specified after the filter name and\nan equal sign, and are separated from each other by a colon.\n\nThere exist so-called source filters that do not have an audio/video input, and sink filters\nthat will not have audio/video output.\n", "subsections": [] }, "GRAPH": { "content": "The graph2dot program included in the FFmpeg tools directory can be used to parse a\nfiltergraph description and issue a corresponding textual representation in the dot language.\n\nInvoke the command:\n\ngraph2dot -h\n\nto see how to use graph2dot.\n\nYou can then pass the dot description to the dot program (from the graphviz suite of\nprograms) and obtain a graphical representation of the filtergraph.\n\nFor example the sequence of commands:\n\necho | \\\ntools/graph2dot -o graph.tmp && \\\ndot -Tpng graph.tmp -o graph.png && \\\ndisplay graph.png\n\ncan be used to create and display an image representing the graph described by the\nGRAPHDESCRIPTION string. Note that this string must be a complete self-contained graph, with\nits inputs and outputs explicitly defined. For example if your command line is of the form:\n\nffmpeg -i infile -vf scale=640:360 outfile\n\nyour GRAPHDESCRIPTION string will need to be of the form:\n\nnullsrc,scale=640:360,nullsink\n\nyou may also need to set the nullsrc parameters and add a format filter in order to simulate\na specific input file.\n", "subsections": [] }, "FILTERGRAPH DESCRIPTION": { "content": "A filtergraph is a directed graph of connected filters. It can contain cycles, and there can\nbe multiple links between a pair of filters. Each link has one input pad on one side\nconnecting it to one filter from which it takes its input, and one output pad on the other\nside connecting it to one filter accepting its output.\n\nEach filter in a filtergraph is an instance of a filter class registered in the application,\nwhich defines the features and the number of input and output pads of the filter.\n\nA filter with no input pads is called a \"source\", and a filter with no output pads is called\na \"sink\".\n", "subsections": [ { "name": "Filtergraph syntax", "content": "A filtergraph has a textual representation, which is recognized by the -filter/-vf/-af and" }, { "name": "-filter -vf -af", "content": "\"avfiltergraphparseptr()\" function defined in libavfilter/avfilter.h.\n\nA filterchain consists of a sequence of connected filters, each one connected to the previous\none in the sequence. A filterchain is represented by a list of \",\"-separated filter\ndescriptions.\n\nA filtergraph consists of a sequence of filterchains. A sequence of filterchains is\nrepresented by a list of \";\"-separated filterchain descriptions.\n\nA filter is represented by a string of the form:\n[inlink1]...[inlinkN]filtername@id=arguments[outlink1]...[outlinkM]\n\nfiltername is the name of the filter class of which the described filter is an instance of,\nand has to be the name of one of the filter classes registered in the program optionally\nfollowed by \"@id\". The name of the filter class is optionally followed by a string\n\"=arguments\".\n\narguments is a string which contains the parameters used to initialize the filter instance.\nIt may have one of two forms:\n\n• A ':'-separated list of key=value pairs.\n\n• A ':'-separated list of value. In this case, the keys are assumed to be the option names\nin the order they are declared. E.g. the \"fade\" filter declares three options in this\norder -- type, startframe and nbframes. Then the parameter list in:0:30 means that the\nvalue in is assigned to the option type, 0 to startframe and 30 to nbframes.\n\n• A ':'-separated list of mixed direct value and long key=value pairs. The direct value\nmust precede the key=value pairs, and follow the same constraints order of the previous\npoint. The following key=value pairs can be set in any preferred order.\n\nIf the option value itself is a list of items (e.g. the \"format\" filter takes a list of pixel\nformats), the items in the list are usually separated by |.\n\nThe list of arguments can be quoted using the character ' as initial and ending mark, and the\ncharacter \\ for escaping the characters within the quoted text; otherwise the argument string\nis considered terminated when the next special character (belonging to the set []=;,) is\nencountered.\n\nThe name and arguments of the filter are optionally preceded and followed by a list of link\nlabels. A link label allows one to name a link and associate it to a filter output or input\npad. The preceding labels inlink1 ... inlinkN, are associated to the filter input pads,\nthe following labels outlink1 ... outlinkM, are associated to the output pads.\n\nWhen two link labels with the same name are found in the filtergraph, a link between the\ncorresponding input and output pad is created.\n\nIf an output pad is not labelled, it is linked by default to the first unlabelled input pad\nof the next filter in the filterchain. For example in the filterchain\n\nnullsrc, split[L1], [L2]overlay, nullsink\n\nthe split filter instance has two output pads, and the overlay filter instance two input\npads. The first output pad of split is labelled \"L1\", the first input pad of overlay is\nlabelled \"L2\", and the second output pad of split is linked to the second input pad of\noverlay, which are both unlabelled.\n\nIn a filter description, if the input label of the first filter is not specified, \"in\" is\nassumed; if the output label of the last filter is not specified, \"out\" is assumed.\n\nIn a complete filterchain all the unlabelled filter input and output pads must be connected.\nA filtergraph is considered valid if all the filter input and output pads of all the\nfilterchains are connected.\n\nLibavfilter will automatically insert scale filters where format conversion is required. It\nis possible to specify swscale flags for those automatically inserted scalers by prepending\n\"swsflags=flags;\" to the filtergraph description.\n\nHere is a BNF description of the filtergraph syntax:\n\n ::= sequence of alphanumeric characters and ''\n ::= [\"@\"]\n ::= \"[\" \"]\"\n ::= []\n ::= sequence of chars (possibly quoted)\n ::= [] [\"=\" ] []\n ::= [,]\n ::= [swsflags=;] [;]\n" }, { "name": "Notes on filtergraph escaping", "content": "Filtergraph description composition entails several levels of escaping. See the \"Quoting and\nescaping\" section in the ffmpeg-utils(1) manual for more information about the employed\nescaping procedure.\n\nA first level escaping affects the content of each filter option value, which may contain the\nspecial character \":\" used to separate values, or one of the escaping characters \"\\'\".\n\nA second level escaping affects the whole filter description, which may contain the escaping\ncharacters \"\\'\" or the special characters \"[],;\" used by the filtergraph description.\n\nFinally, when you specify a filtergraph on a shell commandline, you need to perform a third\nlevel escaping for the shell special characters contained within it.\n\nFor example, consider the following string to be embedded in the drawtext filter description\ntext value:\n\nthis is a 'string': may contain one, or more, special characters\n\nThis string contains the \"'\" special escaping character, and the \":\" special character, so it\nneeds to be escaped in this way:\n\ntext=this is a \\'string\\'\\: may contain one, or more, special characters\n\nA second level of escaping is required when embedding the filter description in a filtergraph\ndescription, in order to escape all the filtergraph special characters. Thus the example\nabove becomes:\n\ndrawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters\n\n(note that in addition to the \"\\'\" escaping special characters, also \",\" needs to be\nescaped).\n\nFinally an additional level of escaping is needed when writing the filtergraph description in\na shell command, which depends on the escaping rules of the adopted shell. For example,\nassuming that \"\\\" is special and needs to be escaped with another \"\\\", the previous string\nwill finally result in:\n\n-vf \"drawtext=text=this is a \\\\\\\\\\\\'string\\\\\\\\\\\\'\\\\\\\\: may contain one\\\\, or more\\\\, special characters\"\n" } ] }, "TIMELINE EDITING": { "content": "Some filters support a generic enable option. For the filters supporting timeline editing,\nthis option can be set to an expression which is evaluated before sending a frame to the\nfilter. If the evaluation is non-zero, the filter will be enabled, otherwise the frame will\nbe sent unchanged to the next filter in the filtergraph.\n\nThe expression accepts the following values:\n\nt timestamp expressed in seconds, NAN if the input timestamp is unknown\n\nn sequential number of the input frame, starting from 0\n\npos the position in the file of the input frame, NAN if unknown\n\nw\nh width and height of the input frame if video\n\nAdditionally, these filters support an enable command that can be used to re-define the\nexpression.\n\nLike any other filtering option, the enable option follows the same rules.\n\nFor example, to enable a blur filter (smartblur) from 10 seconds to 3 minutes, and a curves\nfilter starting at 3 seconds:\n\nsmartblur = enable='between(t,10,3*60)',\ncurves = enable='gte(t,3)' : preset=crossprocess\n\nSee \"ffmpeg -filters\" to view which filters have timeline support.\n", "subsections": [] }, "CHANGING OPTIONS AT RUNTIME WITH A COMMAND": { "content": "Some options can be changed during the operation of the filter using a command. These options\nare marked 'T' on the output of ffmpeg -h filter=. The name of the command\nis the name of the option and the argument is the new value.\n", "subsections": [] }, "OPTIONS FOR FILTERS WITH SEVERAL INPUTS": { "content": "Some filters with several inputs support a common set of options. These options can only be\nset by name, not with the short notation.\n\neofaction\nThe action to take when EOF is encountered on the secondary input; it accepts one of the\nfollowing values:\n\nrepeat\nRepeat the last frame (the default).\n\nendall\nEnd both streams.\n\npass\nPass the main input through.\n", "subsections": [ { "name": "shortest", "content": "If set to 1, force the output to terminate when the shortest input terminates. Default\nvalue is 0.\n" }, { "name": "repeatlast", "content": "If set to 1, force the filter to extend the last frame of secondary streams until the end\nof the primary stream. A value of 0 disables this behavior. Default value is 1.\n" } ] }, "AUDIO FILTERS": { "content": "When you configure your FFmpeg build, you can disable any of the existing filters using\n\"--disable-filters\". The configure output will show the audio filters included in your\nbuild.\n\nBelow is a description of the currently available audio filters.\n", "subsections": [ { "name": "acompressor", "content": "A compressor is mainly used to reduce the dynamic range of a signal. Especially modern music\nis mostly compressed at a high ratio to improve the overall loudness. It's done to get the\nhighest attention of a listener, \"fatten\" the sound and bring more \"power\" to the track. If\na signal is compressed too much it may sound dull or \"dead\" afterwards or it may start to\n\"pump\" (which could be a powerful effect but can also destroy a track completely). The right\ncompression is the key to reach a professional sound and is the high art of mixing and\nmastering. Because of its complex settings it may take a long time to get the right feeling\nfor this kind of effect.\n\nCompression is done by detecting the volume above a chosen level \"threshold\" and dividing it\nby the factor set with \"ratio\". So if you set the threshold to -12dB and your signal reaches\n-6dB a ratio of 2:1 will result in a signal at -9dB. Because an exact manipulation of the\nsignal would cause distortion of the waveform the reduction can be levelled over the time.\nThis is done by setting \"Attack\" and \"Release\". \"attack\" determines how long the signal has\nto rise above the threshold before any reduction will occur and \"release\" sets the time the\nsignal has to fall below the threshold to reduce the reduction again. Shorter signals than\nthe chosen attack time will be left untouched. The overall reduction of the signal can be\nmade up afterwards with the \"makeup\" setting. So compressing the peaks of a signal about 6dB\nand raising the makeup to this level results in a signal twice as loud than the source. To\ngain a softer entry in the compression the \"knee\" flattens the hard edge at the threshold in\nthe range of the chosen decibels.\n\nThe filter accepts the following options:\n\nlevelin\nSet input gain. Default is 1. Range is between 0.015625 and 64.\n" }, { "name": "mode", "content": "Set mode of compressor operation. Can be \"upward\" or \"downward\". Default is \"downward\".\n" }, { "name": "threshold", "content": "If a signal of stream rises above this level it will affect the gain reduction. By\ndefault it is 0.125. Range is between 0.00097563 and 1.\n" }, { "name": "ratio", "content": "Set a ratio by which the signal is reduced. 1:2 means that if the level rose 4dB above\nthe threshold, it will be only 2dB above after the reduction. Default is 2. Range is\nbetween 1 and 20.\n" }, { "name": "attack", "content": "Amount of milliseconds the signal has to rise above the threshold before gain reduction\nstarts. Default is 20. Range is between 0.01 and 2000.\n" }, { "name": "release", "content": "Amount of milliseconds the signal has to fall below the threshold before reduction is\ndecreased again. Default is 250. Range is between 0.01 and 9000.\n" }, { "name": "makeup", "content": "Set the amount by how much signal will be amplified after processing. Default is 1.\nRange is from 1 to 64.\n" }, { "name": "knee", "content": "Curve the sharp knee around the threshold to enter gain reduction more softly. Default\nis 2.82843. Range is between 1 and 8.\n" }, { "name": "link", "content": "Choose if the \"average\" level between all channels of input stream or the\nlouder(\"maximum\") channel of input stream affects the reduction. Default is \"average\".\n" }, { "name": "detection", "content": "Should the exact signal be taken in case of \"peak\" or an RMS one in case of \"rms\".\nDefault is \"rms\" which is mostly smoother.\n\nmix How much to use compressed signal in output. Default is 1. Range is between 0 and 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "acontrast", "content": "Simple audio dynamic range compression/expansion filter.\n\nThe filter accepts the following options:\n" }, { "name": "contrast", "content": "Set contrast. Default is 33. Allowed range is between 0 and 100.\n" }, { "name": "acopy", "content": "Copy the input audio source unchanged to the output. This is mainly useful for testing\npurposes.\n" }, { "name": "acrossfade", "content": "Apply cross fade from one input audio stream to another input audio stream. The cross fade\nis applied for specified duration near the end of first stream.\n\nThe filter accepts the following options:\n\nnbsamples, ns\nSpecify the number of samples for which the cross fade effect has to last. At the end of\nthe cross fade effect the first input audio will be completely silent. Default is 44100.\n" }, { "name": "duration, d", "content": "Specify the duration of the cross fade effect. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax. By default the duration is determined by\nnbsamples. If set this option is used instead of nbsamples.\n" }, { "name": "overlap, o", "content": "Should first stream end overlap with second stream start. Default is enabled.\n" }, { "name": "curve1", "content": "Set curve for cross fade transition for first stream.\n" }, { "name": "curve2", "content": "Set curve for cross fade transition for second stream.\n\nFor description of available curve types see afade filter description.\n\nExamples\n\n• Cross fade from one input to another:\n\nffmpeg -i first.flac -i second.flac -filtercomplex acrossfade=d=10:c1=exp:c2=exp output.flac\n\n• Cross fade from one input to another but without overlapping:\n\nffmpeg -i first.flac -i second.flac -filtercomplex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac\n" }, { "name": "acrossover", "content": "Split audio stream into several bands.\n\nThis filter splits audio stream into two or more frequency ranges. Summing all streams back\nwill give flat output.\n\nThe filter accepts the following options:\n" }, { "name": "split", "content": "Set split frequencies. Those must be positive and increasing.\n" }, { "name": "order", "content": "Set filter order for each band split. This controls filter roll-off or steepness of\nfilter transfer function. Available values are:\n\n2nd 12 dB per octave.\n\n4th 24 dB per octave.\n\n6th 36 dB per octave.\n\n8th 48 dB per octave.\n\n10th\n60 dB per octave.\n\n12th\n72 dB per octave.\n\n14th\n84 dB per octave.\n\n16th\n96 dB per octave.\n\n18th\n108 dB per octave.\n\n20th\n120 dB per octave.\n\nDefault is 4th.\n" }, { "name": "level", "content": "Set input gain level. Allowed range is from 0 to 1. Default value is 1.\n" }, { "name": "gains", "content": "Set output gain for each band. Default value is 1 for all bands.\n\nExamples\n\n• Split input audio stream into two bands (low and high) with split frequency of 1500 Hz,\neach band will be in separate stream:\n\nffmpeg -i in.flac -filtercomplex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav\n\n• Same as above, but with higher filter order:\n\nffmpeg -i in.flac -filtercomplex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav\n\n• Same as above, but also with additional middle band (frequencies between 1500 and 8000):\n\nffmpeg -i in.flac -filtercomplex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav\n" }, { "name": "acrusher", "content": "Reduce audio bit resolution.\n\nThis filter is bit crusher with enhanced functionality. A bit crusher is used to audibly\nreduce number of bits an audio signal is sampled with. This doesn't change the bit depth at\nall, it just produces the effect. Material reduced in bit depth sounds more harsh and\n\"digital\". This filter is able to even round to continuous values instead of discrete bit\ndepths. Additionally it has a D/C offset which results in different crushing of the lower\nand the upper half of the signal. An Anti-Aliasing setting is able to produce \"softer\"\ncrushing sounds.\n\nAnother feature of this filter is the logarithmic mode. This setting switches from linear\ndistances between bits to logarithmic ones. The result is a much more \"natural\" sounding\ncrusher which doesn't gate low signals for example. The human ear has a logarithmic\nperception, so this kind of crushing is much more pleasant. Logarithmic crushing is also\nable to get anti-aliased.\n\nThe filter accepts the following options:\n\nlevelin\nSet level in.\n\nlevelout\nSet level out.\n" }, { "name": "bits", "content": "Set bit reduction.\n\nmix Set mixing amount.\n" }, { "name": "mode", "content": "Can be linear: \"lin\" or logarithmic: \"log\".\n\ndc Set DC.\n\naa Set anti-aliasing.\n" }, { "name": "samples", "content": "Set sample reduction.\n\nlfo Enable LFO. By default disabled.\n" }, { "name": "lforange", "content": "Set LFO range.\n" }, { "name": "lforate", "content": "Set LFO rate.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "acue", "content": "Delay audio filtering until a given wallclock timestamp. See the cue filter.\n" }, { "name": "adeclick", "content": "Remove impulsive noise from input audio.\n\nSamples detected as impulsive noise are replaced by interpolated samples using autoregressive\nmodelling.\n" }, { "name": "window, w", "content": "Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55\nmilliseconds. This sets size of window which will be processed at once.\n" }, { "name": "overlap, o", "content": "Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default\nvalue is 75 percent. Setting this to a very high value increases impulsive noise removal\nbut makes whole process much slower.\n" }, { "name": "arorder, a", "content": "Set autoregression order, in percentage of window size. Allowed range is from 0 to 25.\nDefault value is 2 percent. This option also controls quality of interpolated samples\nusing neighbour good samples.\n" }, { "name": "threshold, t", "content": "Set threshold value. Allowed range is from 1 to 100. Default value is 2. This controls\nthe strength of impulsive noise which is going to be removed. The lower value, the more\nsamples will be detected as impulsive noise.\n" }, { "name": "burst, b", "content": "Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default value\nis 2. If any two samples detected as noise are spaced less than this value then any\nsample between those two samples will be also detected as noise.\n" }, { "name": "method, m", "content": "Set overlap method.\n\nIt accepts the following values:\n\nadd, a\nSelect overlap-add method. Even not interpolated samples are slightly changed with\nthis method.\n\nsave, s\nSelect overlap-save method. Not interpolated samples remain unchanged.\n\nDefault value is \"a\".\n" }, { "name": "adeclip", "content": "Remove clipped samples from input audio.\n\nSamples detected as clipped are replaced by interpolated samples using autoregressive\nmodelling.\n" }, { "name": "window, w", "content": "Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55\nmilliseconds. This sets size of window which will be processed at once.\n" }, { "name": "overlap, o", "content": "Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default\nvalue is 75 percent.\n" }, { "name": "arorder, a", "content": "Set autoregression order, in percentage of window size. Allowed range is from 0 to 25.\nDefault value is 8 percent. This option also controls quality of interpolated samples\nusing neighbour good samples.\n" }, { "name": "threshold, t", "content": "Set threshold value. Allowed range is from 1 to 100. Default value is 10. Higher values\nmake clip detection less aggressive.\n" }, { "name": "hsize, n", "content": "Set size of histogram used to detect clips. Allowed range is from 100 to 9999. Default\nvalue is 1000. Higher values make clip detection less aggressive.\n" }, { "name": "method, m", "content": "Set overlap method.\n\nIt accepts the following values:\n\nadd, a\nSelect overlap-add method. Even not interpolated samples are slightly changed with\nthis method.\n\nsave, s\nSelect overlap-save method. Not interpolated samples remain unchanged.\n\nDefault value is \"a\".\n" }, { "name": "adelay", "content": "Delay one or more audio channels.\n\nSamples in delayed channel are filled with silence.\n\nThe filter accepts the following option:\n" }, { "name": "delays", "content": "Set list of delays in milliseconds for each channel separated by '|'. Unused delays will\nbe silently ignored. If number of given delays is smaller than number of channels all\nremaining channels will not be delayed. If you want to delay exact number of samples,\nappend 'S' to number. If you want instead to delay in seconds, append 's' to number.\n\nall Use last set delay for all remaining channels. By default is disabled. This option if\nenabled changes how option \"delays\" is interpreted.\n\nExamples\n\n• Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave the second\nchannel (and any other channels that may be present) unchanged.\n\nadelay=1500|0|500\n\n• Delay second channel by 500 samples, the third channel by 700 samples and leave the first\nchannel (and any other channels that may be present) unchanged.\n\nadelay=0|500S|700S\n\n• Delay all channels by same number of samples:\n\nadelay=delays=64S:all=1\n" }, { "name": "adenorm", "content": "Remedy denormals in audio by adding extremely low-level noise.\n\nThis filter shall be placed before any filter that can produce denormals.\n\nA description of the accepted parameters follows.\n" }, { "name": "level", "content": "Set level of added noise in dB. Default is \"-351\". Allowed range is from -451 to -90.\n" }, { "name": "type", "content": "Set type of added noise.\n\ndc Add DC signal.\n\nac Add AC signal.\n\nsquare\nAdd square signal.\n\npulse\nAdd pulse signal.\n\nDefault is \"dc\".\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "aderivative, aintegral", "content": "Compute derivative/integral of audio stream.\n\nApplying both filters one after another produces original audio.\n" }, { "name": "aecho", "content": "Apply echoing to the input audio.\n\nEchoes are reflected sound and can occur naturally amongst mountains (and sometimes large\nbuildings) when talking or shouting; digital echo effects emulate this behaviour and are\noften used to help fill out the sound of a single instrument or vocal. The time difference\nbetween the original signal and the reflection is the \"delay\", and the loudness of the\nreflected signal is the \"decay\". Multiple echoes can have different delays and decays.\n\nA description of the accepted parameters follows.\n\ningain\nSet input gain of reflected signal. Default is 0.6.\n\noutgain\nSet output gain of reflected signal. Default is 0.3.\n" }, { "name": "delays", "content": "Set list of time intervals in milliseconds between original signal and reflections\nseparated by '|'. Allowed range for each \"delay\" is \"(0 - 90000.0]\". Default is 1000.\n" }, { "name": "decays", "content": "Set list of loudness of reflected signals separated by '|'. Allowed range for each\n\"decay\" is \"(0 - 1.0]\". Default is 0.5.\n\nExamples\n\n• Make it sound as if there are twice as many instruments as are actually playing:\n\naecho=0.8:0.88:60:0.4\n\n• If delay is very short, then it sounds like a (metallic) robot playing music:\n\naecho=0.8:0.88:6:0.4\n\n• A longer delay will sound like an open air concert in the mountains:\n\naecho=0.8:0.9:1000:0.3\n\n• Same as above but with one more mountain:\n\naecho=0.8:0.9:1000|1800:0.3|0.25\n" }, { "name": "aemphasis", "content": "Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs\nwith different filter curves. E.g. to store music on vinyl the signal has to be altered by a\nfilter first to even out the disadvantages of this recording medium. Once the material is\nplayed back the inverse filter has to be applied to restore the distortion of the frequency\nresponse.\n\nThe filter accepts the following options:\n\nlevelin\nSet input gain.\n\nlevelout\nSet output gain.\n" }, { "name": "mode", "content": "Set filter mode. For restoring material use \"reproduction\" mode, otherwise use\n\"production\" mode. Default is \"reproduction\" mode.\n" }, { "name": "type", "content": "Set filter type. Selects medium. Can be one of the following:\n\ncol select Columbia.\n\nemi select EMI.\n\nbsi select BSI (78RPM).\n\nriaa\nselect RIAA.\n\ncd select Compact Disc (CD).\n\n50fm\nselect 50Xs (FM).\n\n75fm\nselect 75Xs (FM).\n\n50kf\nselect 50Xs (FM-KF).\n\n75kf\nselect 75Xs (FM-KF).\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "aeval", "content": "Modify an audio signal according to the specified expressions.\n\nThis filter accepts one or more expressions (one for each channel), which are evaluated and\nused to modify a corresponding audio signal.\n\nIt accepts the following parameters:\n" }, { "name": "exprs", "content": "Set the '|'-separated expressions list for each separate channel. If the number of input\nchannels is greater than the number of expressions, the last specified expression is used\nfor the remaining output channels.\n\nchannellayout, c\nSet output channel layout. If not specified, the channel layout is specified by the\nnumber of expressions. If set to same, it will use by default the same input channel\nlayout.\n\nEach expression in exprs can contain the following constants and functions:\n\nch channel number of the current expression\n\nn number of the evaluated sample, starting from 0\n\ns sample rate\n\nt time of the evaluated sample expressed in seconds\n\nnbinchannels\nnboutchannels\ninput and output number of channels\n" }, { "name": "val(CH)", "content": "the value of input channel with number CH\n\nNote: this filter is slow. For faster processing you should use a dedicated filter.\n\nExamples\n\n• Half volume:\n\naeval=val(ch)/2:c=same\n\n• Invert phase of the second channel:\n\naeval=val(0)|-val(1)\n" }, { "name": "aexciter", "content": "An exciter is used to produce high sound that is not present in the original signal. This is\ndone by creating harmonic distortions of the signal which are restricted in range and added\nto the original signal. An Exciter raises the upper end of an audio signal without simply\nraising the higher frequencies like an equalizer would do to create a more \"crisp\" or\n\"brilliant\" sound.\n\nThe filter accepts the following options:\n\nlevelin\nSet input level prior processing of signal. Allowed range is from 0 to 64. Default\nvalue is 1.\n\nlevelout\nSet output level after processing of signal. Allowed range is from 0 to 64. Default\nvalue is 1.\n" }, { "name": "amount", "content": "Set the amount of harmonics added to original signal. Allowed range is from 0 to 64.\nDefault value is 1.\n" }, { "name": "drive", "content": "Set the amount of newly created harmonics. Allowed range is from 0.1 to 10. Default\nvalue is 8.5.\n" }, { "name": "blend", "content": "Set the octave of newly created harmonics. Allowed range is from -10 to 10. Default\nvalue is 0.\n" }, { "name": "freq", "content": "Set the lower frequency limit of producing harmonics in Hz. Allowed range is from 2000\nto 12000 Hz. Default is 7500 Hz.\n" }, { "name": "ceil", "content": "Set the upper frequency limit of producing harmonics. Allowed range is from 9999 to\n20000 Hz. If value is lower than 10000 Hz no limit is applied.\n" }, { "name": "listen", "content": "Mute the original signal and output only added harmonics. By default is disabled.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "afade", "content": "Apply fade-in/out effect to input audio.\n\nA description of the accepted parameters follows.\n" }, { "name": "type, t", "content": "Specify the effect type, can be either \"in\" for fade-in, or \"out\" for a fade-out effect.\nDefault is \"in\".\n\nstartsample, ss\nSpecify the number of the start sample for starting to apply the fade effect. Default is\n0.\n\nnbsamples, ns\nSpecify the number of samples for which the fade effect has to last. At the end of the\nfade-in effect the output audio will have the same volume as the input audio, at the end\nof the fade-out transition the output audio will be silence. Default is 44100.\n\nstarttime, st\nSpecify the start time of the fade effect. Default is 0. The value must be specified as\na time duration; see the Time duration section in the ffmpeg-utils(1) manual for the\naccepted syntax. If set this option is used instead of startsample.\n" }, { "name": "duration, d", "content": "Specify the duration of the fade effect. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax. At the end of the fade-in effect the\noutput audio will have the same volume as the input audio, at the end of the fade-out\ntransition the output audio will be silence. By default the duration is determined by\nnbsamples. If set this option is used instead of nbsamples.\n" }, { "name": "curve", "content": "Set curve for fade transition.\n\nIt accepts the following values:\n\ntri select triangular, linear slope (default)\n\nqsin\nselect quarter of sine wave\n\nhsin\nselect half of sine wave\n\nesin\nselect exponential sine wave\n\nlog select logarithmic\n\nipar\nselect inverted parabola\n\nqua select quadratic\n\ncub select cubic\n\nsqu select square root\n\ncbr select cubic root\n\npar select parabola\n\nexp select exponential\n\niqsin\nselect inverted quarter of sine wave\n\nihsin\nselect inverted half of sine wave\n\ndese\nselect double-exponential seat\n\ndesi\nselect double-exponential sigmoid\n\nlosi\nselect logistic sigmoid\n\nsinc\nselect sine cardinal function\n\nisinc\nselect inverted sine cardinal function\n\nnofade\nno fade applied\n\nCommands\n\nThis filter supports the all above options as commands.\n\nExamples\n\n• Fade in first 15 seconds of audio:\n\nafade=t=in:ss=0:d=15\n\n• Fade out last 25 seconds of a 900 seconds audio:\n\nafade=t=out:st=875:d=25\n" }, { "name": "afftdn", "content": "Denoise audio samples with FFT.\n\nA description of the accepted parameters follows.\n\nnr Set the noise reduction in dB, allowed range is 0.01 to 97. Default value is 12 dB.\n\nnf Set the noise floor in dB, allowed range is -80 to -20. Default value is -50 dB.\n\nnt Set the noise type.\n\nIt accepts the following values:\n\nw Select white noise.\n\nv Select vinyl noise.\n\ns Select shellac noise.\n\nc Select custom noise, defined in \"bn\" option.\n\nDefault value is white noise.\n\nbn Set custom band noise for every one of 15 bands. Bands are separated by ' ' or '|'.\n\nrf Set the residual floor in dB, allowed range is -80 to -20. Default value is -38 dB.\n\ntn Enable noise tracking. By default is disabled. With this enabled, noise floor is\nautomatically adjusted.\n\ntr Enable residual tracking. By default is disabled.\n\nom Set the output mode.\n\nIt accepts the following values:\n\ni Pass input unchanged.\n\no Pass noise filtered out.\n\nn Pass only noise.\n\nDefault value is o.\n\nCommands\n\nThis filter supports the following commands:\n\nsamplenoise, sn\nStart or stop measuring noise profile. Syntax for the command is : \"start\" or \"stop\"\nstring. After measuring noise profile is stopped it will be automatically applied in\nfiltering.\n\nnoisereduction, nr\nChange noise reduction. Argument is single float number. Syntax for the command is :\n\"noisereduction\"\n\nnoisefloor, nf\nChange noise floor. Argument is single float number. Syntax for the command is :\n\"noisefloor\"\n\noutputmode, om\nChange output mode operation. Syntax for the command is : \"i\", \"o\" or \"n\" string.\n" }, { "name": "afftfilt", "content": "Apply arbitrary expressions to samples in frequency domain.\n" }, { "name": "real", "content": "Set frequency domain real expression for each separate channel separated by '|'. Default\nis \"re\". If the number of input channels is greater than the number of expressions, the\nlast specified expression is used for the remaining output channels.\n" }, { "name": "imag", "content": "Set frequency domain imaginary expression for each separate channel separated by '|'.\nDefault is \"im\".\n\nEach expression in real and imag can contain the following constants and functions:\n\nsr sample rate\n\nb current frequency bin number\n\nnb number of available bins\n\nch channel number of the current expression\n\nchs number of channels\n\npts current frame pts\n\nre current real part of frequency bin of current channel\n\nim current imaginary part of frequency bin of current channel\n\nreal(b, ch)\nReturn the value of real part of frequency bin at location (bin,channel)\n\nimag(b, ch)\nReturn the value of imaginary part of frequency bin at location (bin,channel)\n\nwinsize\nSet window size. Allowed range is from 16 to 131072. Default is 4096\n\nwinfunc\nSet window function. Default is \"hann\".\n" }, { "name": "overlap", "content": "Set window overlap. If set to 1, the recommended overlap for selected window function\nwill be picked. Default is 0.75.\n\nExamples\n\n• Leave almost only low frequencies in audio:\n\nafftfilt=\"'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'\"\n\n• Apply robotize effect:\n\nafftfilt=\"real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':winsize=512:overlap=0.75\"\n\n• Apply whisper effect:\n\nafftfilt=\"real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':winsize=128:overlap=0.8\"\n" }, { "name": "afir", "content": "Apply an arbitrary Finite Impulse Response filter.\n\nThis filter is designed for applying long FIR filters, up to 60 seconds long.\n\nIt can be used as component for digital crossover filters, room equalization, cross talk\ncancellation, wavefield synthesis, auralization, ambiophonics, ambisonics and spatialization.\n\nThis filter uses the streams higher than first one as FIR coefficients. If the non-first\nstream holds a single channel, it will be used for all input channels in the first stream,\notherwise the number of channels in the non-first stream must be same as the number of\nchannels in the first stream.\n\nIt accepts the following parameters:\n\ndry Set dry gain. This sets input gain.\n\nwet Set wet gain. This sets final output gain.\n" }, { "name": "length", "content": "Set Impulse Response filter length. Default is 1, which means whole IR is processed.\n" }, { "name": "gtype", "content": "Enable applying gain measured from power of IR.\n\nSet which approach to use for auto gain measurement.\n\nnone\nDo not apply any gain.\n\npeak\nselect peak gain, very conservative approach. This is default value.\n\ndc select DC gain, limited application.\n\ngn select gain to noise approach, this is most popular one.\n" }, { "name": "irgain", "content": "Set gain to be applied to IR coefficients before filtering. Allowed range is 0 to 1.\nThis gain is applied after any gain applied with gtype option.\n" }, { "name": "irfmt", "content": "Set format of IR stream. Can be \"mono\" or \"input\". Default is \"input\".\n" }, { "name": "maxir", "content": "Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds.\nAllowed range is 0.1 to 60 seconds.\n" }, { "name": "response", "content": "Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in\nadditional video stream. By default it is disabled.\n" }, { "name": "channel", "content": "Set for which IR channel to display frequency response. By default is first channel\ndisplayed. This option is used only when response is enabled.\n" }, { "name": "size", "content": "Set video stream size. This option is used only when response is enabled.\n" }, { "name": "rate", "content": "Set video stream frame rate. This option is used only when response is enabled.\n" }, { "name": "minp", "content": "Set minimal partition size used for convolution. Default is 8192. Allowed range is from\n1 to 32768. Lower values decreases latency at cost of higher CPU usage.\n" }, { "name": "maxp", "content": "Set maximal partition size used for convolution. Default is 8192. Allowed range is from\n8 to 32768. Lower values may increase CPU usage.\n" }, { "name": "nbirs", "content": "Set number of input impulse responses streams which will be switchable at runtime.\nAllowed range is from 1 to 32. Default is 1.\n\nir Set IR stream which will be used for convolution, starting from 0, should always be lower\nthan supplied value by \"nbirs\" option. Default is 0. This option can be changed at\nruntime via commands.\n\nExamples\n\n• Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:\n\nffmpeg -i input.wav -i middletunnel1waymono.wav -lavfi afir output.wav\n" }, { "name": "aformat", "content": "Set output format constraints for the input audio. The framework will negotiate the most\nappropriate format to minimize conversions.\n\nIt accepts the following parameters:\n\nsamplefmts, f\nA '|'-separated list of requested sample formats.\n\nsamplerates, r\nA '|'-separated list of requested sample rates.\n\nchannellayouts, cl\nA '|'-separated list of requested channel layouts.\n\nSee the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.\n\nIf a parameter is omitted, all values are allowed.\n\nForce the output to either unsigned 8-bit or signed 16-bit stereo\n\naformat=samplefmts=u8|s16:channellayouts=stereo\n" }, { "name": "afreqshift", "content": "Apply frequency shift to input audio samples.\n\nThe filter accepts the following options:\n" }, { "name": "shift", "content": "Specify frequency shift. Allowed range is -INTMAX to INTMAX. Default value is 0.0.\n" }, { "name": "level", "content": "Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value\nis 1.0.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "agate", "content": "A gate is mainly used to reduce lower parts of a signal. This kind of signal processing\nreduces disturbing noise between useful signals.\n\nGating is done by detecting the volume below a chosen level threshold and dividing it by the\nfactor set with ratio. The bottom of the noise floor is set via range. Because an exact\nmanipulation of the signal would cause distortion of the waveform the reduction can be\nlevelled over time. This is done by setting attack and release.\n\nattack determines how long the signal has to fall below the threshold before any reduction\nwill occur and release sets the time the signal has to rise above the threshold to reduce the\nreduction again. Shorter signals than the chosen attack time will be left untouched.\n\nlevelin\nSet input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.\n" }, { "name": "mode", "content": "Set the mode of operation. Can be \"upward\" or \"downward\". Default is \"downward\". If set\nto \"upward\" mode, higher parts of signal will be amplified, expanding dynamic range in\nupward direction. Otherwise, in case of \"downward\" lower parts of signal will be\nreduced.\n" }, { "name": "range", "content": "Set the level of gain reduction when the signal is below the threshold. Default is\n0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then\nfilter behaves like expander.\n" }, { "name": "threshold", "content": "If a signal rises above this level the gain reduction is released. Default is 0.125.\nAllowed range is from 0 to 1.\n" }, { "name": "ratio", "content": "Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to\n9000.\n" }, { "name": "attack", "content": "Amount of milliseconds the signal has to rise above the threshold before gain reduction\nstops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.\n" }, { "name": "release", "content": "Amount of milliseconds the signal has to fall below the threshold before the reduction is\nincreased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.\n" }, { "name": "makeup", "content": "Set amount of amplification of signal after processing. Default is 1. Allowed range is\nfrom 1 to 64.\n" }, { "name": "knee", "content": "Curve the sharp knee around the threshold to enter gain reduction more softly. Default\nis 2.828427125. Allowed range is from 1 to 8.\n" }, { "name": "detection", "content": "Choose if exact signal should be taken for detection or an RMS like one. Default is\n\"rms\". Can be \"peak\" or \"rms\".\n" }, { "name": "link", "content": "Choose if the average level between all channels or the louder channel affects the\nreduction. Default is \"average\". Can be \"average\" or \"maximum\".\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "aiir", "content": "Apply an arbitrary Infinite Impulse Response filter.\n\nIt accepts the following parameters:\n" }, { "name": "zeros, z", "content": "Set B/numerator/zeros/reflection coefficients.\n" }, { "name": "poles, p", "content": "Set A/denominator/poles/ladder coefficients.\n" }, { "name": "gains, k", "content": "Set channels gains.\n\ndrygain\nSet input gain.\n\nwetgain\nSet output gain.\n" }, { "name": "format, f", "content": "Set coefficients format.\n\nll lattice-ladder function\n\nsf analog transfer function\n\ntf digital transfer function\n\nzp Z-plane zeros/poles, cartesian (default)\n\npr Z-plane zeros/poles, polar radians\n\npd Z-plane zeros/poles, polar degrees\n\nsp S-plane zeros/poles\n" }, { "name": "process, r", "content": "Set type of processing.\n\nd direct processing\n\ns serial processing\n\np parallel processing\n" }, { "name": "precision, e", "content": "Set filtering precision.\n\ndbl double-precision floating-point (default)\n\nflt single-precision floating-point\n\ni32 32-bit integers\n\ni16 16-bit integers\n" }, { "name": "normalize, n", "content": "Normalize filter coefficients, by default is enabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n\nmix How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "response", "content": "Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in\nadditional video stream. By default it is disabled.\n" }, { "name": "channel", "content": "Set for which IR channel to display frequency response. By default is first channel\ndisplayed. This option is used only when response is enabled.\n" }, { "name": "size", "content": "Set video stream size. This option is used only when response is enabled.\n\nCoefficients in \"tf\" and \"sf\" format are separated by spaces and are in ascending order.\n\nCoefficients in \"zp\" format are separated by spaces and order of coefficients doesn't matter.\nCoefficients in \"zp\" format are complex numbers with i imaginary unit.\n\nDifferent coefficients and gains can be provided for every channel, in such case use '|' to\nseparate coefficients or gains. Last provided coefficients will be used for all remaining\nchannels.\n\nExamples\n\n• Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample rate:\n\naiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d\n\n• Same as above but in \"zp\" format:\n\naiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s\n\n• Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer\nfunction format:\n\naiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d\n" }, { "name": "alimiter", "content": "The limiter prevents an input signal from rising over a desired threshold. This limiter uses\nlookahead technology to prevent your signal from distorting. It means that there is a small\ndelay after the signal is processed. Keep in mind that the delay it produces is the attack\ntime you set.\n\nThe filter accepts the following options:\n\nlevelin\nSet input gain. Default is 1.\n\nlevelout\nSet output gain. Default is 1.\n" }, { "name": "limit", "content": "Don't let signals above this level pass the limiter. Default is 1.\n" }, { "name": "attack", "content": "The limiter will reach its attenuation level in this amount of time in milliseconds.\nDefault is 5 milliseconds.\n" }, { "name": "release", "content": "Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50\nmilliseconds.\n\nasc When gain reduction is always needed ASC takes care of releasing to an average reduction\nlevel rather than reaching a reduction of 0 in the release time.\n\nasclevel\nSelect how much the release time is affected by ASC, 0 means nearly no changes in release\ntime while 1 produces higher release times.\n" }, { "name": "level", "content": "Auto level output signal. Default is enabled. This normalizes audio back to 0dB if\nenabled.\n\nDepending on picked setting it is recommended to upsample input 2x or 4x times with aresample\nbefore applying this filter.\n" }, { "name": "allpass", "content": "Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width\nwidth. An all-pass filter changes the audio's frequency to phase relationship without\nchanging its frequency to amplitude relationship.\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set frequency in Hz.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Specify the band-width of a filter in widthtype units.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "order, o", "content": "Set the filter order, can be 1 or 2. Default is 2.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change allpass frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange allpass widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change allpass width. Syntax for the command is : \"width\"\n" }, { "name": "mix, m", "content": "Change allpass mix. Syntax for the command is : \"mix\"\n" }, { "name": "aloop", "content": "Loop audio samples.\n\nThe filter accepts the following options:\n" }, { "name": "loop", "content": "Set the number of loops. Setting this value to -1 will result in infinite loops. Default\nis 0.\n" }, { "name": "size", "content": "Set maximal number of samples. Default is 0.\n" }, { "name": "start", "content": "Set first sample of loop. Default is 0.\n" }, { "name": "amerge", "content": "Merge two or more audio streams into a single multi-channel stream.\n\nThe filter accepts the following options:\n" }, { "name": "inputs", "content": "Set the number of inputs. Default is 2.\n\nIf the channel layouts of the inputs are disjoint, and therefore compatible, the channel\nlayout of the output will be set accordingly and the channels will be reordered as necessary.\nIf the channel layouts of the inputs are not disjoint, the output will have all the channels\nof the first input then all the channels of the second input, in that order, and the channel\nlayout of the output will be the default value corresponding to the total number of channels.\n\nFor example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then\nthe output will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3\n(a1 is the first channel of the first input, b1 is the first channel of the second input).\n\nOn the other hand, if both input are in stereo, the output channels will be in the default\norder: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or\nmay not be the expected value.\n\nAll inputs must have the same sample rate, and format.\n\nIf inputs do not have the same duration, the output will stop with the shortest.\n\nExamples\n\n• Merge two mono files into a stereo stream:\n\namovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge\n\n• Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:\n\nffmpeg -i input.mkv -filtercomplex \"[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6\" -c:a pcms16le output.mkv\n" }, { "name": "amix", "content": "Mixes multiple audio inputs into a single output.\n\nNote that this filter only supports float samples (the amerge and pan audio filters support\nmany formats). If the amix input has integer samples then aresample will be automatically\ninserted to perform the conversion to float samples.\n\nFor example\n\nffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filtercomplex amix=inputs=3:duration=first:dropouttransition=3 OUTPUT\n\nwill mix 3 input audio streams to a single output with the same duration as the first input\nand a dropout transition time of 3 seconds.\n\nIt accepts the following parameters:\n" }, { "name": "inputs", "content": "The number of inputs. If unspecified, it defaults to 2.\n" }, { "name": "duration", "content": "How to determine the end-of-stream.\n\nlongest\nThe duration of the longest input. (default)\n\nshortest\nThe duration of the shortest input.\n\nfirst\nThe duration of the first input.\n\ndropouttransition\nThe transition time, in seconds, for volume renormalization when an input stream ends.\nThe default value is 2 seconds.\n" }, { "name": "weights", "content": "Specify weight of each input audio stream as sequence. Each weight is separated by\nspace. By default all inputs have same weight.\n" }, { "name": "normalize", "content": "Always scale inputs instead of only doing summation of samples. Beware of heavy clipping\nif inputs are not normalized prior or after filtering by this filter if this option is\ndisabled. By default is enabled.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "weights", "content": "sum Syntax is same as option with same name.\n" }, { "name": "amultiply", "content": "Multiply first audio stream with second audio stream and store result in output audio stream.\nMultiplication is done by multiplying each sample from first stream with sample at same\nposition from second stream.\n\nWith this element-wise multiplication one can create amplitude fades and amplitude\nmodulations.\n" }, { "name": "anequalizer", "content": "High-order parametric multiband equalizer for each channel.\n\nIt accepts the following parameters:\n" }, { "name": "params", "content": "This option string is in format: \"cchn f=cf w=w g=g t=f | ...\" Each equalizer band is\nseparated by '|'.\n\nchn Set channel number to which equalization will be applied. If input doesn't have that\nchannel the entry is ignored.\n\nf Set central frequency for band. If input doesn't have that frequency the entry is\nignored.\n\nw Set band width in Hertz.\n\ng Set band gain in dB.\n\nt Set filter type for band, optional, can be:\n\n0 Butterworth, this is default.\n\n1 Chebyshev type 1.\n\n2 Chebyshev type 2.\n" }, { "name": "curves", "content": "With this option activated frequency response of anequalizer is displayed in video\nstream.\n" }, { "name": "size", "content": "Set video stream size. Only useful if curves option is activated.\n" }, { "name": "mgain", "content": "Set max gain that will be displayed. Only useful if curves option is activated. Setting\nthis to a reasonable value makes it possible to display gain which is derived from\nneighbour bands which are too close to each other and thus produce higher gain when both\nare activated.\n" }, { "name": "fscale", "content": "Set frequency scale used to draw frequency response in video output. Can be linear or\nlogarithmic. Default is logarithmic.\n" }, { "name": "colors", "content": "Set color for each channel curve which is going to be displayed in video stream. This is\nlist of color names separated by space or by '|'. Unrecognised or missing colors will be\nreplaced by white color.\n\nExamples\n\n• Lower gain by 10 of central frequency 200Hz and width 100 Hz for first 2 channels using\nChebyshev type 1 filter:\n\nanequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "change", "content": "Alter existing filter parameters. Syntax for the commands is :\n\"fN|f=freq|w=width|g=gain\"\n\nfN is existing filter number, starting from 0, if no such filter is available error is\nreturned. freq set new frequency parameter. width set new width parameter in Hertz.\ngain set new gain parameter in dB.\n\nFull filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer\nchange 0|f=200|w=50|g=1',anequalizer=...\n" }, { "name": "anlmdn", "content": "Reduce broadband noise in audio samples using Non-Local Means algorithm.\n\nEach sample is adjusted by looking for other samples with similar contexts. This context\nsimilarity is defined by comparing their surrounding patches of size p. Patches are searched\nin an area of r around the sample.\n\nThe filter accepts the following options:\n\ns Set denoising strength. Allowed range is from 0.00001 to 10. Default value is 0.00001.\n\np Set patch radius duration. Allowed range is from 1 to 100 milliseconds. Default value is\n2 milliseconds.\n\nr Set research radius duration. Allowed range is from 2 to 300 milliseconds. Default value\nis 6 milliseconds.\n\no Set the output mode.\n\nIt accepts the following values:\n\ni Pass input unchanged.\n\no Pass noise filtered out.\n\nn Pass only noise.\n\nDefault value is o.\n\nm Set smooth factor. Default value is 11. Allowed range is from 1 to 15.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "anlms", "content": "Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second\naudio stream.\n\nThis adaptive filter is used to mimic a desired filter by finding the filter coefficients\nthat relate to producing the least mean square of the error signal (difference between the\ndesired, 2nd input audio stream and the actual signal, the 1st input audio stream).\n\nA description of the accepted options follows.\n" }, { "name": "order", "content": "Set filter order.\n\nmu Set filter mu.\n\neps Set the filter eps.\n" }, { "name": "leakage", "content": "Set the filter leakage.\n\noutmode\nIt accepts the following values:\n\ni Pass the 1st input.\n\nd Pass the 2nd input.\n\no Pass filtered samples.\n\nn Pass difference between desired and filtered samples.\n\nDefault value is o.\n\nExamples\n\n• One of many usages of this filter is noise reduction, input audio is filtered with same\nsamples that are delayed by fixed amount, one such example for stereo audio is:\n\nasplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:outmode=o\n\nCommands\n\nThis filter supports the same commands as options, excluding option \"order\".\n" }, { "name": "anull", "content": "Pass the audio source unchanged to the output.\n" }, { "name": "apad", "content": "Pad the end of an audio stream with silence.\n\nThis can be used together with ffmpeg -shortest to extend audio streams to the same length as\nthe video stream.\n\nA description of the accepted options follows.\n\npacketsize\nSet silence packet size. Default value is 4096.\n\npadlen\nSet the number of samples of silence to add to the end. After the value is reached, the\nstream is terminated. This option is mutually exclusive with wholelen.\n\nwholelen\nSet the minimum total number of samples in the output audio stream. If the value is\nlonger than the input audio length, silence is added to the end, until the value is\nreached. This option is mutually exclusive with padlen.\n\npaddur\nSpecify the duration of samples of silence to add. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-zero value.\n\nwholedur\nSpecify the minimum total duration in the output audio stream. See the Time duration\nsection in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-\nzero value. If the value is longer than the input audio length, silence is added to the\nend, until the value is reached. This option is mutually exclusive with paddur\n\nIf neither the padlen nor the wholelen nor paddur nor wholedur option is set, the filter\nwill add silence to the end of the input stream indefinitely.\n\nExamples\n\n• Add 1024 samples of silence to the end of the input:\n\napad=padlen=1024\n\n• Make sure the audio output will contain at least 10000 samples, pad the input with\nsilence if required:\n\napad=wholelen=10000\n\n• Use ffmpeg to pad the audio input with silence, so that the video stream will always\nresult the shortest and will be converted until the end in the output file when using the\nshortest option:\n\nffmpeg -i VIDEO -i AUDIO -filtercomplex \"[1:0]apad\" -shortest OUTPUT\n" }, { "name": "aphaser", "content": "Add a phasing effect to the input audio.\n\nA phaser filter creates series of peaks and troughs in the frequency spectrum. The position\nof the peaks and troughs are modulated so that they vary over time, creating a sweeping\neffect.\n\nA description of the accepted parameters follows.\n\ningain\nSet input gain. Default is 0.4.\n\noutgain\nSet output gain. Default is 0.74\n" }, { "name": "delay", "content": "Set delay in milliseconds. Default is 3.0.\n" }, { "name": "decay", "content": "Set decay. Default is 0.4.\n" }, { "name": "speed", "content": "Set modulation speed in Hz. Default is 0.5.\n" }, { "name": "type", "content": "Set modulation type. Default is triangular.\n\nIt accepts the following values:\n\ntriangular, t\nsinusoidal, s\n" }, { "name": "aphaseshift", "content": "Apply phase shift to input audio samples.\n\nThe filter accepts the following options:\n" }, { "name": "shift", "content": "Specify phase shift. Allowed range is from -1.0 to 1.0. Default value is 0.0.\n" }, { "name": "level", "content": "Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value\nis 1.0.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "apulsator", "content": "Audio pulsator is something between an autopanner and a tremolo. But it can produce funny\nstereo effects as well. Pulsator changes the volume of the left and right channel based on a\nLFO (low frequency oscillator) with different waveforms and shifted phases. This filter have\nthe ability to define an offset between left and right channel. An offset of 0 means that\nboth LFO shapes match each other. The left and right channel are altered equally - a\nconventional tremolo. An offset of 50% means that the shape of the right channel is exactly\nshifted in phase (or moved backwards about half of the frequency) - pulsator acts as an\nautopanner. At 1 both curves match again. Every setting in between moves the phase shift\ngapless between all stages and produces some \"bypassing\" sounds with sine and triangle\nwaveforms. The more you set the offset near 1 (starting from the 0.5) the faster the signal\npasses from the left to the right speaker.\n\nThe filter accepts the following options:\n\nlevelin\nSet input gain. By default it is 1. Range is [0.015625 - 64].\n\nlevelout\nSet output gain. By default it is 1. Range is [0.015625 - 64].\n" }, { "name": "mode", "content": "Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or\nsawdown. Default is sine.\n" }, { "name": "amount", "content": "Set modulation. Define how much of original signal is affected by the LFO.\n\noffsetl\nSet left channel offset. Default is 0. Allowed range is [0 - 1].\n\noffsetr\nSet right channel offset. Default is 0.5. Allowed range is [0 - 1].\n" }, { "name": "width", "content": "Set pulse width. Default is 1. Allowed range is [0 - 2].\n" }, { "name": "timing", "content": "Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.\n\nbpm Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to bpm.\n\nms Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to ms.\n\nhz Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing is\nset to hz.\n" }, { "name": "aresample", "content": "Resample the input audio to the specified parameters, using the libswresample library. If\nnone are specified then the filter will automatically convert between its input and output.\n\nThis filter is also able to stretch/squeeze the audio data to make it match the timestamps or\nto inject silence / cut out audio to make it match the timestamps, do a combination of both\nor do neither.\n\nThe filter accepts the syntax [samplerate:]resampleroptions, where samplerate expresses a\nsample rate and resampleroptions is a list of key=value pairs, separated by \":\". See the\n\"Resampler Options\" section in the ffmpeg-resampler(1) manual for the complete list of\nsupported options.\n\nExamples\n\n• Resample the input audio to 44100Hz:\n\naresample=44100\n\n• Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per\nsecond compensation:\n\naresample=async=1000\n" }, { "name": "areverse", "content": "Reverse an audio clip.\n\nWarning: This filter requires memory to buffer the entire clip, so trimming is suggested.\n\nExamples\n\n• Take the first 5 seconds of a clip, and reverse it.\n\natrim=end=5,areverse\n" }, { "name": "arnndn", "content": "Reduce noise from speech using Recurrent Neural Networks.\n\nThis filter accepts the following options:\n" }, { "name": "model, m", "content": "Set train model file to load. This option is always required.\n\nmix Set how much to mix filtered samples into final output. Allowed range is from -1 to 1.\nDefault value is 1. Negative values are special, they set how much to keep filtered\nnoise in the final filter output. Set this option to -1 to hear actual noise removed from\ninput signal.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "asetnsamples", "content": "Set the number of samples per each output audio frame.\n\nThe last output packet may contain a different number of samples, as the filter will flush\nall the remaining samples when the input audio signals its end.\n\nThe filter accepts the following options:\n\nnboutsamples, n\nSet the number of frames per each output audio frame. The number is intended as the\nnumber of samples per each channel. Default value is 1024.\n" }, { "name": "pad, p", "content": "If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame\nwill contain the same number of samples as the previous ones. Default value is 1.\n\nFor example, to set the number of per-frame samples to 1234 and disable padding for the last\nframe, use:\n\nasetnsamples=n=1234:p=0\n" }, { "name": "asetrate", "content": "Set the sample rate without altering the PCM data. This will result in a change of speed and\npitch.\n\nThe filter accepts the following options:\n\nsamplerate, r\nSet the output sample rate. Default is 44100 Hz.\n" }, { "name": "ashowinfo", "content": "Show a line containing various information for each input audio frame. The input audio is\nnot modified.\n\nThe shown line contains a sequence of key/value pairs of the form key:value.\n\nThe following values are shown in the output:\n\nn The (sequential) number of the input frame, starting from 0.\n\npts The presentation timestamp of the input frame, in time base units; the time base depends\non the filter input pad, and is usually 1/samplerate.\n\nptstime\nThe presentation timestamp of the input frame in seconds.\n\npos position of the frame in the input stream, -1 if this information in unavailable and/or\nmeaningless (for example in case of synthetic audio)\n\nfmt The sample format.\n" }, { "name": "chlayout", "content": "The channel layout.\n" }, { "name": "rate", "content": "The sample rate for the audio frame.\n\nnbsamples\nThe number of samples (per channel) in the frame.\n" }, { "name": "checksum", "content": "The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio, the\ndata is treated as if all the planes were concatenated.\n\nplanechecksums\nA list of Adler-32 checksums for each data plane.\n" }, { "name": "asoftclip", "content": "Apply audio soft clipping.\n\nSoft clipping is a type of distortion effect where the amplitude of a signal is saturated\nalong a smooth curve, rather than the abrupt shape of hard-clipping.\n\nThis filter accepts the following options:\n" }, { "name": "type", "content": "Set type of soft-clipping.\n\nIt accepts the following values:\n\nhard\ntanh\natan\ncubic\nexp\nalg\nquintic\nsin\nerf" }, { "name": "threshold", "content": "Set threshold from where to start clipping. Default value is 0dB or 1.\n" }, { "name": "output", "content": "Set gain applied to output. Default value is 0dB or 1.\n" }, { "name": "param", "content": "Set additional parameter which controls sigmoid function.\n" }, { "name": "oversample", "content": "Set oversampling factor.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "asr", "content": "Automatic Speech Recognition\n\nThis filter uses PocketSphinx for speech recognition. To enable compilation of this filter,\nyou need to configure FFmpeg with \"--enable-pocketsphinx\".\n\nIt accepts the following options:\n" }, { "name": "rate", "content": "Set sampling rate of input audio. Defaults is 16000. This need to match speech models,\notherwise one will get poor results.\n\nhmm Set dictionary containing acoustic model files.\n" }, { "name": "dict", "content": "Set pronunciation dictionary.\n\nlm Set language model file.\n" }, { "name": "lmctl", "content": "Set language model set.\n" }, { "name": "lmname", "content": "Set which language model to use.\n" }, { "name": "logfn", "content": "Set output for log messages.\n\nThe filter exports recognized speech as the frame metadata \"lavfi.asr.text\".\n" }, { "name": "astats", "content": "Display time domain statistical information about the audio channels. Statistics are\ncalculated and displayed for each audio channel and, where applicable, an overall figure is\nalso given.\n\nIt accepts the following option:\n" }, { "name": "length", "content": "Short window length in seconds, used for peak and trough RMS measurement. Default is\n0.05 (50 milliseconds). Allowed range is \"[0.01 - 10]\".\n" }, { "name": "metadata", "content": "Set metadata injection. All the metadata keys are prefixed with \"lavfi.astats.X\", where\n\"X\" is channel number starting from 1 or string \"Overall\". Default is disabled.\n\nAvailable keys for each channel are: DCoffset Minlevel Maxlevel Mindifference\nMaxdifference Meandifference RMSdifference Peaklevel RMSpeak RMStrough Crestfactor\nFlatfactor Peakcount Noisefloor Noisefloorcount Bitdepth Dynamicrange\nZerocrossings Zerocrossingsrate NumberofNaNs NumberofInfs Numberofdenormals\n\nand for Overall: DCoffset Minlevel Maxlevel Mindifference Maxdifference\nMeandifference RMSdifference Peaklevel RMSlevel RMSpeak RMStrough Flatfactor\nPeakcount Noisefloor Noisefloorcount Bitdepth Numberofsamples NumberofNaNs\nNumberofInfs Numberofdenormals\n\nFor example full key look like this \"lavfi.astats.1.DCoffset\" or this\n\"lavfi.astats.Overall.Peakcount\".\n\nFor description what each key means read below.\n" }, { "name": "reset", "content": "Set number of frame after which stats are going to be recalculated. Default is disabled.\n\nmeasureperchannel\nSelect the entries which need to be measured per channel. The metadata keys can be used\nas flags, default is all which measures everything. none disables all per channel\nmeasurement.\n\nmeasureoverall\nSelect the entries which need to be measured overall. The metadata keys can be used as\nflags, default is all which measures everything. none disables all overall measurement.\n\nA description of each shown parameter follows:\n" }, { "name": "DC offset", "content": "Mean amplitude displacement from zero.\n" }, { "name": "Min level", "content": "Minimal sample level.\n" }, { "name": "Max level", "content": "Maximal sample level.\n" }, { "name": "Min difference", "content": "Minimal difference between two consecutive samples.\n" }, { "name": "Max difference", "content": "Maximal difference between two consecutive samples.\n" }, { "name": "Mean difference", "content": "Mean difference between two consecutive samples. The average of each difference between\ntwo consecutive samples.\n" }, { "name": "RMS difference", "content": "Root Mean Square difference between two consecutive samples.\n" }, { "name": "Peak level dB", "content": "" }, { "name": "RMS level dB", "content": "Standard peak and RMS level measured in dBFS.\n" }, { "name": "RMS peak dB", "content": "" }, { "name": "RMS trough dB", "content": "Peak and trough values for RMS level measured over a short window.\n" }, { "name": "Crest factor", "content": "Standard ratio of peak to RMS level (note: not in dB).\n" }, { "name": "Flat factor", "content": "Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels\n(i.e. either Min level or Max level).\n" }, { "name": "Peak count", "content": "Number of occasions (not the number of samples) that the signal attained either Min level\nor Max level.\n" }, { "name": "Noise floor dB", "content": "Minimum local peak measured in dBFS over a short window.\n" }, { "name": "Noise floor count", "content": "Number of occasions (not the number of samples) that the signal attained Noise floor.\n" }, { "name": "Bit depth", "content": "Overall bit depth of audio. Number of bits used for each sample.\n" }, { "name": "Dynamic range", "content": "Measured dynamic range of audio in dB.\n" }, { "name": "Zero crossings", "content": "Number of points where the waveform crosses the zero level axis.\n" }, { "name": "Zero crossings rate", "content": "Rate of Zero crossings and number of audio samples.\n" }, { "name": "asubboost", "content": "Boost subwoofer frequencies.\n\nThe filter accepts the following options:\n\ndry Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1. Default\nvalue is 0.7.\n\nwet Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1. Default\nvalue is 0.7.\n" }, { "name": "decay", "content": "Set delay line decay gain value. Allowed range is from 0 to 1. Default value is 0.7.\n" }, { "name": "feedback", "content": "Set delay line feedback gain value. Allowed range is from 0 to 1. Default value is 0.9.\n" }, { "name": "cutoff", "content": "Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default value is 100.\n" }, { "name": "slope", "content": "Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1. Default value is\n0.5.\n" }, { "name": "delay", "content": "Set delay. Allowed range is from 1 to 100. Default value is 20.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "asubcut", "content": "Cut subwoofer frequencies.\n\nThis filter allows to set custom, steeper roll off than highpass filter, and thus is able to\nmore attenuate frequency content in stop-band.\n\nThe filter accepts the following options:\n" }, { "name": "cutoff", "content": "Set cutoff frequency in Hertz. Allowed range is 2 to 200. Default value is 20.\n" }, { "name": "order", "content": "Set filter order. Available values are from 3 to 20. Default value is 10.\n" }, { "name": "level", "content": "Set input gain level. Allowed range is from 0 to 1. Default value is 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "asupercut", "content": "Cut super frequencies.\n\nThe filter accepts the following options:\n" }, { "name": "cutoff", "content": "Set cutoff frequency in Hertz. Allowed range is 20000 to 192000. Default value is 20000.\n" }, { "name": "order", "content": "Set filter order. Available values are from 3 to 20. Default value is 10.\n" }, { "name": "level", "content": "Set input gain level. Allowed range is from 0 to 1. Default value is 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "asuperpass", "content": "Apply high order Butterworth band-pass filter.\n\nThe filter accepts the following options:\n" }, { "name": "centerf", "content": "Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.\n" }, { "name": "order", "content": "Set filter order. Available values are from 4 to 20. Default value is 4.\n" }, { "name": "qfactor", "content": "Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.\n" }, { "name": "level", "content": "Set input gain level. Allowed range is from 0 to 2. Default value is 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "asuperstop", "content": "Apply high order Butterworth band-stop filter.\n\nThe filter accepts the following options:\n" }, { "name": "centerf", "content": "Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.\n" }, { "name": "order", "content": "Set filter order. Available values are from 4 to 20. Default value is 4.\n" }, { "name": "qfactor", "content": "Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.\n" }, { "name": "level", "content": "Set input gain level. Allowed range is from 0 to 2. Default value is 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "atempo", "content": "Adjust audio tempo.\n\nThe filter accepts exactly one parameter, the audio tempo. If not specified then the filter\nwill assume nominal 1.0 tempo. Tempo must be in the [0.5, 100.0] range.\n\nNote that tempo greater than 2 will skip some samples rather than blend them in. If for any\nreason this is a concern it is always possible to daisy-chain several instances of atempo to\nachieve the desired product tempo.\n\nExamples\n\n• Slow down audio to 80% tempo:\n\natempo=0.8\n\n• To speed up audio to 300% tempo:\n\natempo=3\n\n• To speed up audio to 300% tempo by daisy-chaining two atempo instances:\n\natempo=sqrt(3),atempo=sqrt(3)\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "tempo", "content": "Change filter tempo scale factor. Syntax for the command is : \"tempo\"\n" }, { "name": "atrim", "content": "Trim the input so that the output contains one continuous subpart of the input.\n\nIt accepts the following parameters:\n" }, { "name": "start", "content": "Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with\nthe timestamp start will be the first sample in the output.\n\nend Specify time of the first audio sample that will be dropped, i.e. the audio sample\nimmediately preceding the one with the timestamp end will be the last sample in the\noutput.\n\nstartpts\nSame as start, except this option sets the start timestamp in samples instead of seconds.\n\nendpts\nSame as end, except this option sets the end timestamp in samples instead of seconds.\n" }, { "name": "duration", "content": "The maximum duration of the output in seconds.\n\nstartsample\nThe number of the first sample that should be output.\n\nendsample\nThe number of the first sample that should be dropped.\n\nstart, end, and duration are expressed as time duration specifications; see the Time duration\nsection in the ffmpeg-utils(1) manual.\n\nNote that the first two sets of the start/end options and the duration option look at the\nframe timestamp, while the sample options simply count the samples that pass through the\nfilter. So start/endpts and start/endsample will give different results when the timestamps\nare wrong, inexact or do not start at zero. Also note that this filter does not modify the\ntimestamps. If you wish to have the output timestamps start at zero, insert the asetpts\nfilter after the atrim filter.\n\nIf multiple start or end options are set, this filter tries to be greedy and keep all samples\nthat match at least one of the specified constraints. To keep only the part that matches all\nthe constraints at once, chain multiple atrim filters.\n\nThe defaults are such that all the input is kept. So it is possible to set e.g. just the end\nvalues to keep everything before the specified time.\n\nExamples:\n\n• Drop everything except the second minute of input:\n\nffmpeg -i INPUT -af atrim=60:120\n\n• Keep only the first 1000 samples:\n\nffmpeg -i INPUT -af atrim=endsample=1000\n" }, { "name": "axcorrelate", "content": "Calculate normalized cross-correlation between two input audio streams.\n\nResulted samples are always between -1 and 1 inclusive. If result is 1 it means two input\nsamples are highly correlated in that selected segment. Result 0 means they are not\ncorrelated at all. If result is -1 it means two input samples are out of phase, which means\nthey cancel each other.\n\nThe filter accepts the following options:\n" }, { "name": "size", "content": "Set size of segment over which cross-correlation is calculated. Default is 256. Allowed\nrange is from 2 to 131072.\n" }, { "name": "algo", "content": "Set algorithm for cross-correlation. Can be \"slow\" or \"fast\". Default is \"slow\". Fast\nalgorithm assumes mean values over any given segment are always zero and thus need much\nless calculations to make. This is generally not true, but is valid for typical audio\nstreams.\n\nExamples\n\n• Calculate correlation between channels in stereo audio stream:\n\nffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav\n" }, { "name": "bandpass", "content": "Apply a two-pole Butterworth band-pass filter with central frequency frequency, and\n(3dB-point) band-width width. The csg option selects a constant skirt gain (peak gain = Q)\ninstead of the default: constant 0dB peak gain. The filter roll off at 6dB per octave (20dB\nper decade).\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set the filter's central frequency. Default is 3000.\n\ncsg Constant skirt gain if set to 1. Defaults to 0.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Specify the band-width of a filter in widthtype units.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change bandpass frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange bandpass widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change bandpass width. Syntax for the command is : \"width\"\n" }, { "name": "mix, m", "content": "Change bandpass mix. Syntax for the command is : \"mix\"\n" }, { "name": "bandreject", "content": "Apply a two-pole Butterworth band-reject filter with central frequency frequency, and\n(3dB-point) band-width width. The filter roll off at 6dB per octave (20dB per decade).\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set the filter's central frequency. Default is 3000.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Specify the band-width of a filter in widthtype units.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change bandreject frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange bandreject widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change bandreject width. Syntax for the command is : \"width\"\n" }, { "name": "mix, m", "content": "Change bandreject mix. Syntax for the command is : \"mix\"\n" }, { "name": "bass, lowshelf", "content": "Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving filter with\na response similar to that of a standard hi-fi's tone-controls. This is also known as\nshelving equalisation (EQ).\n\nThe filter accepts the following options:\n" }, { "name": "gain, g", "content": "Give the gain at 0 Hz. Its useful range is about -20 (for a large cut) to +20 (for a\nlarge boost). Beware of clipping when using a positive gain.\n" }, { "name": "frequency, f", "content": "Set the filter's central frequency and so can be used to extend or reduce the frequency\nrange to be boosted or cut. The default value is 100 Hz.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Determine how steep is the filter's shelf transition.\n" }, { "name": "poles, p", "content": "Set number of poles. Default is 2.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change bass frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange bass widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change bass width. Syntax for the command is : \"width\"\n" }, { "name": "gain, g", "content": "Change bass gain. Syntax for the command is : \"gain\"\n" }, { "name": "mix, m", "content": "Change bass mix. Syntax for the command is : \"mix\"\n" }, { "name": "biquad", "content": "Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are\nthe numerator and denominator coefficients respectively. and channels, c specify which\nchannels to filter, by default all available are filtered.\n\nCommands\n\nThis filter supports the following commands:\n\na0\na1\na2\nb0\nb1\nb2 Change biquad parameter. Syntax for the command is : \"value\"\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n" }, { "name": "bs2b", "content": "Bauer stereo to binaural transformation, which improves headphone listening of stereo audio\nrecords.\n\nTo enable compilation of this filter you need to configure FFmpeg with \"--enable-libbs2b\".\n\nIt accepts the following parameters:\n" }, { "name": "profile", "content": "Pre-defined crossfeed level.\n\ndefault\nDefault level (fcut=700, feed=50).\n\ncmoy\nChu Moy circuit (fcut=700, feed=60).\n\njmeier\nJan Meier circuit (fcut=650, feed=95).\n" }, { "name": "fcut", "content": "Cut frequency (in Hz).\n" }, { "name": "feed", "content": "Feed level (in Hz).\n" }, { "name": "channelmap", "content": "Remap input channels to new locations.\n\nIt accepts the following parameters:\n\nmap Map channels from input to output. The argument is a '|'-separated list of mappings, each\nin the \"inchannel-outchannel\" or inchannel form. inchannel can be either the name of\nthe input channel (e.g. FL for front left) or its index in the input channel layout.\noutchannel is the name of the output channel or its index in the output channel layout.\nIf outchannel is not given then it is implicitly an index, starting with zero and\nincreasing by one for each mapping.\n\nchannellayout\nThe channel layout of the output stream.\n\nIf no mapping is present, the filter will implicitly map input channels to output channels,\npreserving indices.\n\nExamples\n\n• For example, assuming a 5.1+downmix input MOV file,\n\nffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav\n\nwill create an output WAV file tagged as stereo from the downmix channels of the input.\n\n• To fix a 5.1 WAV improperly encoded in AAC's native channel order\n\nffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav\n" }, { "name": "channelsplit", "content": "Split each channel from an input audio stream into a separate output stream.\n\nIt accepts the following parameters:\n\nchannellayout\nThe channel layout of the input stream. The default is \"stereo\".\n" }, { "name": "channels", "content": "A channel layout describing the channels to be extracted as separate output streams or\n\"all\" to extract each input channel as a separate stream. The default is \"all\".\n\nChoosing channels not present in channel layout in the input will result in an error.\n\nExamples\n\n• For example, assuming a stereo input MP3 file,\n\nffmpeg -i in.mp3 -filtercomplex channelsplit out.mkv\n\nwill create an output Matroska file with two audio streams, one containing only the left\nchannel and the other the right channel.\n\n• Split a 5.1 WAV file into per-channel files:\n\nffmpeg -i in.wav -filtercomplex\n'channelsplit=channellayout=5.1[FL][FR][FC][LFE][SL][SR]'\n-map '[FL]' frontleft.wav -map '[FR]' frontright.wav -map '[FC]'\nfrontcenter.wav -map '[LFE]' lfe.wav -map '[SL]' sideleft.wav -map '[SR]'\nsideright.wav\n\n• Extract only LFE from a 5.1 WAV file:\n\nffmpeg -i in.wav -filtercomplex 'channelsplit=channellayout=5.1:channels=LFE[LFE]'\n-map '[LFE]' lfe.wav\n" }, { "name": "chorus", "content": "Add a chorus effect to the audio.\n\nCan make a single vocal sound like a chorus, but can also be applied to instrumentation.\n\nChorus resembles an echo effect with a short delay, but whereas with echo the delay is\nconstant, with chorus, it is varied using using sinusoidal or triangular modulation. The\nmodulation depth defines the range the modulated delay is played before or after the delay.\nHence the delayed sound will sound slower or faster, that is the delayed sound tuned around\nthe original one, like in a chorus where some vocals are slightly off key.\n\nIt accepts the following parameters:\n\ningain\nSet input gain. Default is 0.4.\n\noutgain\nSet output gain. Default is 0.4.\n" }, { "name": "delays", "content": "Set delays. A typical delay is around 40ms to 60ms.\n" }, { "name": "decays", "content": "Set decays.\n" }, { "name": "speeds", "content": "Set speeds.\n" }, { "name": "depths", "content": "Set depths.\n\nExamples\n\n• A single delay:\n\nchorus=0.7:0.9:55:0.4:0.25:2\n\n• Two delays:\n\nchorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3\n\n• Fuller sounding chorus with three delays:\n\nchorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3\n" }, { "name": "compand", "content": "Compress or expand the audio's dynamic range.\n\nIt accepts the following parameters:\n" }, { "name": "attacks", "content": "" }, { "name": "decays", "content": "A list of times in seconds for each channel over which the instantaneous level of the\ninput signal is averaged to determine its volume. attacks refers to increase of volume\nand decays refers to decrease of volume. For most situations, the attack time (response\nto the audio getting louder) should be shorter than the decay time, because the human ear\nis more sensitive to sudden loud audio than sudden soft audio. A typical value for attack\nis 0.3 seconds and a typical value for decay is 0.8 seconds. If specified number of\nattacks & decays is lower than number of channels, the last set attack/decay will be used\nfor all remaining channels.\n" }, { "name": "points", "content": "A list of points for the transfer function, specified in dB relative to the maximum\npossible signal amplitude. Each key points list must be defined using the following\nsyntax: \"x0/y0|x1/y1|x2/y2|....\" or \"x0/y0 x1/y1 x2/y2 ....\"\n\nThe input values must be in strictly increasing order but the transfer function does not\nhave to be monotonically rising. The point \"0/0\" is assumed but may be overridden (by\n\"0/out-dBn\"). Typical values for the transfer function are \"-70/-70|-60/-20|1/0\".\n" }, { "name": "soft-knee", "content": "Set the curve radius in dB for all joints. It defaults to 0.01.\n" }, { "name": "gain", "content": "Set the additional gain in dB to be applied at all points on the transfer function. This\nallows for easy adjustment of the overall gain. It defaults to 0.\n" }, { "name": "volume", "content": "Set an initial volume, in dB, to be assumed for each channel when filtering starts. This\npermits the user to supply a nominal level initially, so that, for example, a very large\ngain is not applied to initial signal levels before the companding has begun to operate.\nA typical value for audio which is initially quiet is -90 dB. It defaults to 0.\n" }, { "name": "delay", "content": "Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed\nbefore being fed to the volume adjuster. Specifying a delay approximately equal to the\nattack/decay times allows the filter to effectively operate in predictive rather than\nreactive mode. It defaults to 0.\n\nExamples\n\n• Make music with both quiet and loud passages suitable for listening to in a noisy\nenvironment:\n\ncompand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2\n\nAnother example for audio with whisper and explosion parts:\n\ncompand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0\n\n• A noise gate for when the noise is at a lower level than the signal:\n\ncompand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1\n\n• Here is another noise gate, this time for when the noise is at a higher level than the\nsignal (making it, in some ways, similar to squelch):\n\ncompand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1\n\n• 2:1 compression starting at -6dB:\n\ncompand=points=-80/-80|-6/-6|0/-3.8|20/3.5\n\n• 2:1 compression starting at -9dB:\n\ncompand=points=-80/-80|-9/-9|0/-5.3|20/2.9\n\n• 2:1 compression starting at -12dB:\n\ncompand=points=-80/-80|-12/-12|0/-6.8|20/1.9\n\n• 2:1 compression starting at -18dB:\n\ncompand=points=-80/-80|-18/-18|0/-9.8|20/0.7\n\n• 3:1 compression starting at -15dB:\n\ncompand=points=-80/-80|-15/-15|0/-10.8|20/-5.2\n\n• Compressor/Gate:\n\ncompand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6\n\n• Expander:\n\ncompand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3\n\n• Hard limiter at -6dB:\n\ncompand=attacks=0:points=-80/-80|-6/-6|20/-6\n\n• Hard limiter at -12dB:\n\ncompand=attacks=0:points=-80/-80|-12/-12|20/-12\n\n• Hard noise gate at -35 dB:\n\ncompand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20\n\n• Soft limiter:\n\ncompand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8\n" }, { "name": "compensationdelay", "content": "Compensation Delay Line is a metric based delay to compensate differing positions of\nmicrophones or speakers.\n\nFor example, you have recorded guitar with two microphones placed in different locations.\nBecause the front of sound wave has fixed speed in normal conditions, the phasing of\nmicrophones can vary and depends on their location and interposition. The best sound mix can\nbe achieved when these microphones are in phase (synchronized). Note that a distance of ~30\ncm between microphones makes one microphone capture the signal in antiphase to the other\nmicrophone. That makes the final mix sound moody. This filter helps to solve phasing\nproblems by adding different delays to each microphone track and make them synchronized.\n\nThe best result can be reached when you take one track as base and synchronize other tracks\none by one with it. Remember that synchronization/delay tolerance depends on sample rate,\ntoo. Higher sample rates will give more tolerance.\n\nThe filter accepts the following parameters:\n\nmm Set millimeters distance. This is compensation distance for fine tuning. Default is 0.\n\ncm Set cm distance. This is compensation distance for tightening distance setup. Default is\n0.\n\nm Set meters distance. This is compensation distance for hard distance setup. Default is\n0.\n\ndry Set dry amount. Amount of unprocessed (dry) signal. Default is 0.\n\nwet Set wet amount. Amount of processed (wet) signal. Default is 1.\n" }, { "name": "temp", "content": "Set temperature in degrees Celsius. This is the temperature of the environment. Default\nis 20.\n" }, { "name": "crossfeed", "content": "Apply headphone crossfeed filter.\n\nCrossfeed is the process of blending the left and right channels of stereo audio recording.\nIt is mainly used to reduce extreme stereo separation of low frequencies.\n\nThe intent is to produce more speaker like sound to the listener.\n\nThe filter accepts the following options:\n" }, { "name": "strength", "content": "Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1. This sets gain\nof low shelf filter for side part of stereo image. Default is -6dB. Max allowed is -30db\nwhen strength is set to 1.\n" }, { "name": "range", "content": "Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1. This sets cut off\nfrequency of low shelf filter. Default is cut off near 1550 Hz. With range set to 1 cut\noff frequency is set to 2100 Hz.\n" }, { "name": "slope", "content": "Set curve slope of low shelf filter. Default is 0.5. Allowed range is from 0.01 to 1.\n\nlevelin\nSet input gain. Default is 0.9.\n\nlevelout\nSet output gain. Default is 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "crystalizer", "content": "Simple algorithm for audio noise sharpening.\n\nThis filter linearly increases differences betweeen each audio sample.\n\nThe filter accepts the following options:\n\ni Sets the intensity of effect (default: 2.0). Must be in range between -10.0 to 0\n(unchanged sound) to 10.0 (maximum effect). To inverse filtering use negative value.\n\nc Enable clipping. By default is enabled.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "dcshift", "content": "Apply a DC shift to the audio.\n\nThis can be useful to remove a DC offset (caused perhaps by a hardware problem in the\nrecording chain) from the audio. The effect of a DC offset is reduced headroom and hence\nvolume. The astats filter can be used to determine if a signal has a DC offset.\n" }, { "name": "shift", "content": "Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.\n" }, { "name": "limitergain", "content": "Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to\nprevent clipping.\n" }, { "name": "deesser", "content": "Apply de-essing to the audio samples.\n\ni Set intensity for triggering de-essing. Allowed range is from 0 to 1. Default is 0.\n\nm Set amount of ducking on treble part of sound. Allowed range is from 0 to 1. Default is\n0.5.\n\nf How much of original frequency content to keep when de-essing. Allowed range is from 0 to\n1. Default is 0.5.\n\ns Set the output mode.\n\nIt accepts the following values:\n\ni Pass input unchanged.\n\no Pass ess filtered out.\n\ne Pass only ess.\n\nDefault value is o.\n" }, { "name": "drmeter", "content": "Measure audio dynamic range.\n\nDR values of 14 and higher is found in very dynamic material. DR of 8 to 13 is found in\ntransition material. And anything less that 8 have very poor dynamics and is very compressed.\n\nThe filter accepts the following options:\n" }, { "name": "length", "content": "Set window length in seconds used to split audio into segments of equal length. Default\nis 3 seconds.\n" }, { "name": "dynaudnorm", "content": "Dynamic Audio Normalizer.\n\nThis filter applies a certain amount of gain to the input audio in order to bring its peak\nmagnitude to a target level (e.g. 0 dBFS). However, in contrast to more \"simple\"\nnormalization algorithms, the Dynamic Audio Normalizer *dynamically* re-adjusts the gain\nfactor to the input audio. This allows for applying extra gain to the \"quiet\" sections of\nthe audio while avoiding distortions or clipping the \"loud\" sections. In other words: The\nDynamic Audio Normalizer will \"even out\" the volume of quiet and loud sections, in the sense\nthat the volume of each section is brought to the same target level. Note, however, that the\nDynamic Audio Normalizer achieves this goal *without* applying \"dynamic range compressing\".\nIt will retain 100% of the dynamic range *within* each section of the audio file.\n" }, { "name": "framelen, f", "content": "Set the frame length in milliseconds. In range from 10 to 8000 milliseconds. Default is\n500 milliseconds. The Dynamic Audio Normalizer processes the input audio in small\nchunks, referred to as frames. This is required, because a peak magnitude has no meaning\nfor just a single sample value. Instead, we need to determine the peak magnitude for a\ncontiguous sequence of sample values. While a \"standard\" normalizer would simply use the\npeak magnitude of the complete file, the Dynamic Audio Normalizer determines the peak\nmagnitude individually for each frame. The length of a frame is specified in\nmilliseconds. By default, the Dynamic Audio Normalizer uses a frame length of 500\nmilliseconds, which has been found to give good results with most files. Note that the\nexact frame length, in number of samples, will be determined automatically, based on the\nsampling rate of the individual input audio file.\n" }, { "name": "gausssize, g", "content": "Set the Gaussian filter window size. In range from 3 to 301, must be odd number. Default\nis 31. Probably the most important parameter of the Dynamic Audio Normalizer is the\n\"window size\" of the Gaussian smoothing filter. The filter's window size is specified in\nframes, centered around the current frame. For the sake of simplicity, this must be an\nodd number. Consequently, the default value of 31 takes into account the current frame,\nas well as the 15 preceding frames and the 15 subsequent frames. Using a larger window\nresults in a stronger smoothing effect and thus in less gain variation, i.e. slower gain\nadaptation. Conversely, using a smaller window results in a weaker smoothing effect and\nthus in more gain variation, i.e. faster gain adaptation. In other words, the more you\nincrease this value, the more the Dynamic Audio Normalizer will behave like a\n\"traditional\" normalization filter. On the contrary, the more you decrease this value,\nthe more the Dynamic Audio Normalizer will behave like a dynamic range compressor.\n" }, { "name": "peak, p", "content": "Set the target peak value. This specifies the highest permissible magnitude level for the\nnormalized audio input. This filter will try to approach the target peak magnitude as\nclosely as possible, but at the same time it also makes sure that the normalized signal\nwill never exceed the peak magnitude. A frame's maximum local gain factor is imposed\ndirectly by the target peak magnitude. The default value is 0.95 and thus leaves a\nheadroom of 5%*. It is not recommended to go above this value.\n" }, { "name": "maxgain, m", "content": "Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0. The Dynamic\nAudio Normalizer determines the maximum possible (local) gain factor for each input\nframe, i.e. the maximum gain factor that does not result in clipping or distortion. The\nmaximum gain factor is determined by the frame's highest magnitude sample. However, the\nDynamic Audio Normalizer additionally bounds the frame's maximum gain factor by a\npredetermined (global) maximum gain factor. This is done in order to avoid excessive gain\nfactors in \"silent\" or almost silent frames. By default, the maximum gain factor is 10.0,\nFor most inputs the default value should be sufficient and it usually is not recommended\nto increase this value. Though, for input with an extremely low overall volume level, it\nmay be necessary to allow even higher gain factors. Note, however, that the Dynamic Audio\nNormalizer does not simply apply a \"hard\" threshold (i.e. cut off values above the\nthreshold). Instead, a \"sigmoid\" threshold function will be applied. This way, the gain\nfactors will smoothly approach the threshold value, but never exceed that value.\n" }, { "name": "targetrms, r", "content": "Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled. By default, the\nDynamic Audio Normalizer performs \"peak\" normalization. This means that the maximum\nlocal gain factor for each frame is defined (only) by the frame's highest magnitude\nsample. This way, the samples can be amplified as much as possible without exceeding the\nmaximum signal level, i.e. without clipping. Optionally, however, the Dynamic Audio\nNormalizer can also take into account the frame's root mean square, abbreviated RMS. In\nelectrical engineering, the RMS is commonly used to determine the power of a time-varying\nsignal. It is therefore considered that the RMS is a better approximation of the\n\"perceived loudness\" than just looking at the signal's peak magnitude. Consequently, by\nadjusting all frames to a constant RMS value, a uniform \"perceived loudness\" can be\nestablished. If a target RMS value has been specified, a frame's local gain factor is\ndefined as the factor that would result in exactly that RMS value. Note, however, that\nthe maximum local gain factor is still restricted by the frame's highest magnitude\nsample, in order to prevent clipping.\n" }, { "name": "coupling, n", "content": "Enable channels coupling. By default is enabled. By default, the Dynamic Audio\nNormalizer will amplify all channels by the same amount. This means the same gain factor\nwill be applied to all channels, i.e. the maximum possible gain factor is determined by\nthe \"loudest\" channel. However, in some recordings, it may happen that the volume of the\ndifferent channels is uneven, e.g. one channel may be \"quieter\" than the other one(s).\nIn this case, this option can be used to disable the channel coupling. This way, the gain\nfactor will be determined independently for each channel, depending only on the\nindividual channel's highest magnitude sample. This allows for harmonizing the volume of\nthe different channels.\n" }, { "name": "correctdc, c", "content": "Enable DC bias correction. By default is disabled. An audio signal (in the time domain)\nis a sequence of sample values. In the Dynamic Audio Normalizer these sample values are\nrepresented in the -1.0 to 1.0 range, regardless of the original input format. Normally,\nthe audio signal, or \"waveform\", should be centered around the zero point. That means if\nwe calculate the mean value of all samples in a file, or in a single frame, then the\nresult should be 0.0 or at least very close to that value. If, however, there is a\nsignificant deviation of the mean value from 0.0, in either positive or negative\ndirection, this is referred to as a DC bias or DC offset. Since a DC bias is clearly\nundesirable, the Dynamic Audio Normalizer provides optional DC bias correction. With DC\nbias correction enabled, the Dynamic Audio Normalizer will determine the mean value, or\n\"DC correction\" offset, of each input frame and subtract that value from all of the\nframe's sample values which ensures those samples are centered around 0.0 again. Also, in\norder to avoid \"gaps\" at the frame boundaries, the DC correction offset values will be\ninterpolated smoothly between neighbouring frames.\n" }, { "name": "altboundary, b", "content": "Enable alternative boundary mode. By default is disabled. The Dynamic Audio Normalizer\ntakes into account a certain neighbourhood around each frame. This includes the preceding\nframes as well as the subsequent frames. However, for the \"boundary\" frames, located at\nthe very beginning and at the very end of the audio file, not all neighbouring frames are\navailable. In particular, for the first few frames in the audio file, the preceding\nframes are not known. And, similarly, for the last few frames in the audio file, the\nsubsequent frames are not known. Thus, the question arises which gain factors should be\nassumed for the missing frames in the \"boundary\" region. The Dynamic Audio Normalizer\nimplements two modes to deal with this situation. The default boundary mode assumes a\ngain factor of exactly 1.0 for the missing frames, resulting in a smooth \"fade in\" and\n\"fade out\" at the beginning and at the end of the input, respectively.\n" }, { "name": "compress, s", "content": "Set the compress factor. In range from 0.0 to 30.0. Default is 0.0. By default, the\nDynamic Audio Normalizer does not apply \"traditional\" compression. This means that signal\npeaks will not be pruned and thus the full dynamic range will be retained within each\nlocal neighbourhood. However, in some cases it may be desirable to combine the Dynamic\nAudio Normalizer's normalization algorithm with a more \"traditional\" compression. For\nthis purpose, the Dynamic Audio Normalizer provides an optional compression\n(thresholding) function. If (and only if) the compression feature is enabled, all input\nframes will be processed by a soft knee thresholding function prior to the actual\nnormalization process. Put simply, the thresholding function is going to prune all\nsamples whose magnitude exceeds a certain threshold value. However, the Dynamic Audio\nNormalizer does not simply apply a fixed threshold value. Instead, the threshold value\nwill be adjusted for each individual frame. In general, smaller parameters result in\nstronger compression, and vice versa. Values below 3.0 are not recommended, because\naudible distortion may appear.\n" }, { "name": "threshold, t", "content": "Set the target threshold value. This specifies the lowest permissible magnitude level for\nthe audio input which will be normalized. If input frame volume is above this value\nframe will be normalized. Otherwise frame may not be normalized at all. The default\nvalue is set to 0, which means all input frames will be normalized. This option is\nmostly useful if digital noise is not wanted to be amplified.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "earwax", "content": "Make audio easier to listen to on headphones.\n\nThis filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened\nto on headphones the stereo image is moved from inside your head (standard for headphones) to\noutside and in front of the listener (standard for speakers).\n\nPorted from SoX.\n" }, { "name": "equalizer", "content": "Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and\naround a selected frequency can be increased or decreased, whilst (unlike bandpass and\nbandreject filters) that at all other frequencies is unchanged.\n\nIn order to produce complex equalisation curves, this filter can be given several times, each\nwith a different central frequency.\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set the filter's central frequency in Hz.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Specify the band-width of a filter in widthtype units.\n" }, { "name": "gain, g", "content": "Set the required gain or attenuation in dB. Beware of clipping when using a positive\ngain.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nExamples\n\n• Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:\n\nequalizer=f=1000:t=h:width=200:g=-10\n\n• Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:\n\nequalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change equalizer frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange equalizer widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change equalizer width. Syntax for the command is : \"width\"\n" }, { "name": "gain, g", "content": "Change equalizer gain. Syntax for the command is : \"gain\"\n" }, { "name": "mix, m", "content": "Change equalizer mix. Syntax for the command is : \"mix\"\n" }, { "name": "extrastereo", "content": "Linearly increases the difference between left and right channels which adds some sort of\n\"live\" effect to playback.\n\nThe filter accepts the following options:\n\nm Sets the difference coefficient (default: 2.5). 0.0 means mono sound (average of both\nchannels), with 1.0 sound will be unchanged, with -1.0 left and right channels will be\nswapped.\n\nc Enable clipping. By default is enabled.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "firequalizer", "content": "Apply FIR Equalization using arbitrary frequency response.\n\nThe filter accepts the following option:\n" }, { "name": "gain", "content": "Set gain curve equation (in dB). The expression can contain variables:\n\nf the evaluated frequency\n\nsr sample rate\n\nch channel number, set to 0 when multichannels evaluation is disabled\n\nchid\nchannel id, see libavutil/channellayout.h, set to the first channel id when\nmultichannels evaluation is disabled\n\nchs number of channels\n\nchlayout\nchannellayout, see libavutil/channellayout.h\n\nand functions:\n\ngaininterpolate(f)\ninterpolate gain on frequency f based on gainentry\n\ncubicinterpolate(f)\nsame as gaininterpolate, but smoother\n\nThis option is also available as command. Default is gaininterpolate(f).\n\ngainentry\nSet gain entry for gaininterpolate function. The expression can contain functions:\n\nentry(f, g)\nstore gain entry at frequency f with value g\n\nThis option is also available as command.\n" }, { "name": "delay", "content": "Set filter delay in seconds. Higher value means more accurate. Default is 0.01.\n" }, { "name": "accuracy", "content": "Set filter accuracy in Hz. Lower value means more accurate. Default is 5.\n" }, { "name": "wfunc", "content": "Set window function. Acceptable values are:\n\nrectangular\nrectangular window, useful when gain curve is already smooth\n\nhann\nhann window (default)\n\nhamming\nhamming window\n\nblackman\nblackman window\n\nnuttall3\n3-terms continuous 1st derivative nuttall window\n\nmnuttall3\nminimum 3-terms discontinuous nuttall window\n\nnuttall\n4-terms continuous 1st derivative nuttall window\n\nbnuttall\nminimum 4-terms discontinuous nuttall (blackman-nuttall) window\n\nbharris\nblackman-harris window\n\ntukey\ntukey window\n" }, { "name": "fixed", "content": "If enabled, use fixed number of audio samples. This improves speed when filtering with\nlarge delay. Default is disabled.\n" }, { "name": "multi", "content": "Enable multichannels evaluation on gain. Default is disabled.\n\nzerophase\nEnable zero phase mode by subtracting timestamp to compensate delay. Default is\ndisabled.\n" }, { "name": "scale", "content": "Set scale used by gain. Acceptable values are:\n\nlinlin\nlinear frequency, linear gain\n\nlinlog\nlinear frequency, logarithmic (in dB) gain (default)\n\nloglin\nlogarithmic (in octave scale where 20 Hz is 0) frequency, linear gain\n\nloglog\nlogarithmic frequency, logarithmic gain\n" }, { "name": "dumpfile", "content": "Set file for dumping, suitable for gnuplot.\n" }, { "name": "dumpscale", "content": "Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.\n" }, { "name": "fft2", "content": "Enable 2-channel convolution using complex FFT. This improves speed significantly.\nDefault is disabled.\n\nminphase\nEnable minimum phase impulse response. Default is disabled.\n\nExamples\n\n• lowpass at 1000 Hz:\n\nfirequalizer=gain='if(lt(f,1000), 0, -INF)'\n\n• lowpass at 1000 Hz with gainentry:\n\nfirequalizer=gainentry='entry(1000,0); entry(1001, -INF)'\n\n• custom equalization:\n\nfirequalizer=gainentry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'\n\n• higher delay with zero phase to compensate delay:\n\nfirequalizer=delay=0.1:fixed=on:zerophase=on\n\n• lowpass on left channel, highpass on right channel:\n\nfirequalizer=gain='if(eq(chid,1), gaininterpolate(f), if(eq(chid,2), gaininterpolate(1e6+f), 0))'\n:gainentry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on\n" }, { "name": "flanger", "content": "Apply a flanging effect to the audio.\n\nThe filter accepts the following options:\n" }, { "name": "delay", "content": "Set base delay in milliseconds. Range from 0 to 30. Default value is 0.\n" }, { "name": "depth", "content": "Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.\n" }, { "name": "regen", "content": "Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default\nvalue is 0.\n" }, { "name": "width", "content": "Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value\nis 71.\n" }, { "name": "speed", "content": "Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.\n" }, { "name": "shape", "content": "Set swept wave shape, can be triangular or sinusoidal. Default value is sinusoidal.\n" }, { "name": "phase", "content": "Set swept wave percentage-shift for multi channel. Range from 0 to 100. Default value is\n25.\n" }, { "name": "interp", "content": "Set delay-line interpolation, linear or quadratic. Default is linear.\n" }, { "name": "haas", "content": "Apply Haas effect to audio.\n\nNote that this makes most sense to apply on mono signals. With this filter applied to mono\nsignals it give some directionality and stretches its stereo image.\n\nThe filter accepts the following options:\n\nlevelin\nSet input level. By default is 1, or 0dB\n\nlevelout\nSet output level. By default is 1, or 0dB.\n\nsidegain\nSet gain applied to side part of signal. By default is 1.\n\nmiddlesource\nSet kind of middle source. Can be one of the following:\n\nleft\nPick left channel.\n\nright\nPick right channel.\n\nmid Pick middle part signal of stereo image.\n\nside\nPick side part signal of stereo image.\n\nmiddlephase\nChange middle phase. By default is disabled.\n\nleftdelay\nSet left channel delay. By default is 2.05 milliseconds.\n\nleftbalance\nSet left channel balance. By default is -1.\n\nleftgain\nSet left channel gain. By default is 1.\n\nleftphase\nChange left phase. By default is disabled.\n\nrightdelay\nSet right channel delay. By defaults is 2.12 milliseconds.\n\nrightbalance\nSet right channel balance. By default is 1.\n\nrightgain\nSet right channel gain. By default is 1.\n\nrightphase\nChange right phase. By default is enabled.\n" }, { "name": "hdcd", "content": "Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with embedded\nHDCD codes is expanded into a 20-bit PCM stream.\n\nThe filter supports the Peak Extend and Low-level Gain Adjustment features of HDCD, and\ndetects the Transient Filter flag.\n\nffmpeg -i HDCD16.flac -af hdcd OUT24.flac\n\nWhen using the filter with wav, note the default encoding for wav is 16-bit, so the resulting\n20-bit stream will be truncated back to 16-bit. Use something like -acodec pcms24le after\nthe filter to get 24-bit PCM output.\n\nffmpeg -i HDCD16.wav -af hdcd OUT16.wav\nffmpeg -i HDCD16.wav -af hdcd -c:a pcms24le OUT24.wav\n\nThe filter accepts the following options:\n\ndisableautoconvert\nDisable any automatic format conversion or resampling in the filter graph.\n\nprocessstereo\nProcess the stereo channels together. If targetgain does not match between channels,\nconsider it invalid and use the last valid targetgain.\n\ncdtms\nSet the code detect timer period in ms.\n\nforcepe\nAlways extend peaks above -3dBFS even if PE isn't signaled.\n\nanalyzemode\nReplace audio with a solid tone and adjust the amplitude to signal some specific aspect\nof the decoding process. The output file can be loaded in an audio editor alongside the\noriginal to aid analysis.\n\n\"analyzemode=pe:forcepe=true\" can be used to see all samples above the PE level.\n\nModes are:\n\n0, off\nDisabled\n\n1, lle\nGain adjustment level at each sample\n\n2, pe\nSamples where peak extend occurs\n\n3, cdt\nSamples where the code detect timer is active\n\n4, tgm\nSamples where the target gain does not match between channels\n" }, { "name": "headphone", "content": "Apply head-related transfer functions (HRTFs) to create virtual loudspeakers around the user\nfor binaural listening via headphones. The HRIRs are provided via additional streams, for\neach channel one stereo input stream is needed.\n\nThe filter accepts the following options:\n\nmap Set mapping of input streams for convolution. The argument is a '|'-separated list of\nchannel names in order as they are given as additional stream inputs for filter. This\nalso specify number of input streams. Number of input streams must be not less than\nnumber of channels in first stream plus one.\n" }, { "name": "gain", "content": "Set gain applied to audio. Value is in dB. Default is 0.\n" }, { "name": "type", "content": "Set processing type. Can be time or freq. time is processing audio in time domain which\nis slow. freq is processing audio in frequency domain which is fast. Default is freq.\n\nlfe Set custom gain for LFE channels. Value is in dB. Default is 0.\n" }, { "name": "size", "content": "Set size of frame in number of samples which will be processed at once. Default value is\n1024. Allowed range is from 1024 to 96000.\n" }, { "name": "hrir", "content": "Set format of hrir stream. Default value is stereo. Alternative value is multich. If\nvalue is set to stereo, number of additional streams should be greater or equal to number\nof input channels in first input stream. Also each additional stream should have stereo\nnumber of channels. If value is set to multich, number of additional streams should be\nexactly one. Also number of input channels of additional stream should be equal or\ngreater than twice number of channels of first input stream.\n\nExamples\n\n• Full example using wav files as coefficients with amovie filters for 7.1 downmix, each\namovie filter use stereo file with IR coefficients as input. The files give coefficients\nfor each position of virtual loudspeaker:\n\nffmpeg -i input.wav\n-filtercomplex \"amovie=azi270ele0DFC.wav[sr];amovie=azi90ele0DFC.wav[sl];amovie=azi225ele0DFC.wav[br];amovie=azi135ele0DFC.wav[bl];amovie=azi0ele0DFC.wav,asplit[fc][lfe];amovie=azi35ele0DFC.wav[fl];amovie=azi325ele0DFC.wav[fr];[0:a][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR\"\noutput.wav\n\n• Full example using wav files as coefficients with amovie filters for 7.1 downmix, but now\nin multich hrir format.\n\nffmpeg -i input.wav -filtercomplex \"amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich\"\noutput.wav\n" }, { "name": "highpass", "content": "Apply a high-pass filter with 3dB point frequency. The filter can be either single-pole, or\ndouble-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per\ndecade).\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set frequency in Hz. Default is 3000.\n" }, { "name": "poles, p", "content": "Set number of poles. Default is 2.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Specify the band-width of a filter in widthtype units. Applies only to double-pole\nfilter. The default is 0.707q and gives a Butterworth response.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change highpass frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange highpass widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change highpass width. Syntax for the command is : \"width\"\n" }, { "name": "mix, m", "content": "Change highpass mix. Syntax for the command is : \"mix\"\n" }, { "name": "join", "content": "Join multiple input streams into one multi-channel stream.\n\nIt accepts the following parameters:\n" }, { "name": "inputs", "content": "The number of input streams. It defaults to 2.\n\nchannellayout\nThe desired output channel layout. It defaults to stereo.\n\nmap Map channels from inputs to output. The argument is a '|'-separated list of mappings,\neach in the \"inputidx.inchannel-outchannel\" form. inputidx is the 0-based index of\nthe input stream. inchannel can be either the name of the input channel (e.g. FL for\nfront left) or its index in the specified input stream. outchannel is the name of the\noutput channel.\n\nThe filter will attempt to guess the mappings when they are not specified explicitly. It does\nso by first trying to find an unused matching input channel and if that fails it picks the\nfirst unused input channel.\n\nJoin 3 inputs (with properly set channel layouts):\n\nffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filtercomplex join=inputs=3 OUTPUT\n\nBuild a 5.1 output from 6 single-channel streams:\n\nffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filtercomplex\n'join=inputs=6:channellayout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'\nout\n" }, { "name": "ladspa", "content": "Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.\n\nTo enable compilation of this filter you need to configure FFmpeg with \"--enable-ladspa\".\n" }, { "name": "file, f", "content": "Specifies the name of LADSPA plugin library to load. If the environment variable\nLADSPAPATH is defined, the LADSPA plugin is searched in each one of the directories\nspecified by the colon separated list in LADSPAPATH, otherwise in the standard LADSPA\npaths, which are in this order: HOME/.ladspa/lib/, /usr/local/lib/ladspa/,\n/usr/lib/ladspa/.\n" }, { "name": "plugin, p", "content": "Specifies the plugin within the library. Some libraries contain only one plugin, but\nothers contain many of them. If this is not set filter will list all available plugins\nwithin the specified library.\n" }, { "name": "controls, c", "content": "Set the '|' separated list of controls which are zero or more floating point values that\ndetermine the behavior of the loaded plugin (for example delay, threshold or gain).\nControls need to be defined using the following syntax:\nc0=value0|c1=value1|c2=value2|..., where valuei is the value set on the i-th control.\nAlternatively they can be also defined using the following syntax:\nvalue0|value1|value2|..., where valuei is the value set on the i-th control. If controls\nis set to \"help\", all available controls and their valid ranges are printed.\n\nsamplerate, s\nSpecify the sample rate, default to 44100. Only used if plugin have zero inputs.\n\nnbsamples, n\nSet the number of samples per channel per each output frame, default is 1024. Only used\nif plugin have zero inputs.\n" }, { "name": "duration, d", "content": "Set the minimum duration of the sourced audio. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be\ngreater than the specified duration, as the generated audio is always cut at the end of a\ncomplete frame. If not specified, or the expressed duration is negative, the audio is\nsupposed to be generated forever. Only used if plugin have zero inputs.\n" }, { "name": "latency, l", "content": "Enable latency compensation, by default is disabled. Only used if plugin have inputs.\n\nExamples\n\n• List all available plugins within amp (LADSPA example plugin) library:\n\nladspa=file=amp\n\n• List all available controls and their valid ranges for \"vcfnotch\" plugin from \"VCF\"\nlibrary:\n\nladspa=f=vcf:p=vcfnotch:c=help\n\n• Simulate low quality audio equipment using \"Computer Music Toolkit\" (CMT) plugin library:\n\nladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12\n\n• Add reverberation to the audio using TAP-plugins (Tom's Audio Processing plugins):\n\nladspa=file=tapreverb:tapreverb\n\n• Generate white noise, with 0.2 amplitude:\n\nladspa=file=cmt:noisesourcewhite:c=c0=.2\n\n• Generate 20 bpm clicks using plugin \"C* Click - Metronome\" from the \"C* Audio Plugin\nSuite\" (CAPS) library:\n\nladspa=file=caps:Click:c=c1=20'\n\n• Apply \"C* Eq10X2 - Stereo 10-band equaliser\" effect:\n\nladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2\n\n• Increase volume by 20dB using fast lookahead limiter from Steve Harris \"SWH Plugins\"\ncollection:\n\nladspa=fastlookaheadlimiter1913:fastLookaheadLimiter:20|0|2\n\n• Attenuate low frequencies using Multiband EQ from Steve Harris \"SWH Plugins\" collection:\n\nladspa=mbeq1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0\n\n• Reduce stereo image using \"Narrower\" from the \"C* Audio Plugin Suite\" (CAPS) library:\n\nladspa=caps:Narrower\n\n• Another white noise, now using \"C* Audio Plugin Suite\" (CAPS) library:\n\nladspa=caps:White:.2\n\n• Some fractal noise, using \"C* Audio Plugin Suite\" (CAPS) library:\n\nladspa=caps:Fractal:c=c1=1\n\n• Dynamic volume normalization using \"VLevel\" plugin:\n\nladspa=vlevel-ladspa:vlevelmono\n\nCommands\n\nThis filter supports the following commands:\n\ncN Modify the N-th control value.\n\nIf the specified value is not valid, it is ignored and prior one is kept.\n" }, { "name": "loudnorm", "content": "EBU R128 loudness normalization. Includes both dynamic and linear normalization modes.\nSupport for both single pass (livestreams, files) and double pass (files) modes. This\nalgorithm can target IL, LRA, and maximum true peak. In dynamic mode, to accurately detect\ntrue peaks, the audio stream will be upsampled to 192 kHz. Use the \"-ar\" option or\n\"aresample\" filter to explicitly set an output sample rate.\n\nThe filter accepts the following options:\n" }, { "name": "I, i", "content": "Set integrated loudness target. Range is -70.0 - -5.0. Default value is -24.0.\n" }, { "name": "LRA, lra", "content": "Set loudness range target. Range is 1.0 - 20.0. Default value is 7.0.\n" }, { "name": "TP, tp", "content": "Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.\n\nmeasuredI, measuredi\nMeasured IL of input file. Range is -99.0 - +0.0.\n\nmeasuredLRA, measuredlra\nMeasured LRA of input file. Range is 0.0 - 99.0.\n\nmeasuredTP, measuredtp\nMeasured true peak of input file. Range is -99.0 - +99.0.\n\nmeasuredthresh\nMeasured threshold of input file. Range is -99.0 - +0.0.\n" }, { "name": "offset", "content": "Set offset gain. Gain is applied before the true-peak limiter. Range is -99.0 - +99.0.\nDefault is +0.0.\n" }, { "name": "linear", "content": "Normalize by linearly scaling the source audio. \"measuredI\", \"measuredLRA\",\n\"measuredTP\", and \"measuredthresh\" must all be specified. Target LRA shouldn't be lower\nthan source LRA and the change in integrated loudness shouldn't result in a true peak\nwhich exceeds the target TP. If any of these conditions aren't met, normalization mode\nwill revert to dynamic. Options are \"true\" or \"false\". Default is \"true\".\n\ndualmono\nTreat mono input files as \"dual-mono\". If a mono file is intended for playback on a\nstereo system, its EBU R128 measurement will be perceptually incorrect. If set to\n\"true\", this option will compensate for this effect. Multi-channel input files are not\naffected by this option. Options are true or false. Default is false.\n\nprintformat\nSet print format for stats. Options are summary, json, or none. Default value is none.\n" }, { "name": "lowpass", "content": "Apply a low-pass filter with 3dB point frequency. The filter can be either single-pole or\ndouble-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per\ndecade).\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set frequency in Hz. Default is 500.\n" }, { "name": "poles, p", "content": "Set number of poles. Default is 2.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Specify the band-width of a filter in widthtype units. Applies only to double-pole\nfilter. The default is 0.707q and gives a Butterworth response.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nExamples\n\n• Lowpass only LFE channel, it LFE is not present it does nothing:\n\nlowpass=c=LFE\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change lowpass frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange lowpass widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change lowpass width. Syntax for the command is : \"width\"\n" }, { "name": "mix, m", "content": "Change lowpass mix. Syntax for the command is : \"mix\"\n" }, { "name": "lv2", "content": "Load a LV2 (LADSPA Version 2) plugin.\n\nTo enable compilation of this filter you need to configure FFmpeg with \"--enable-lv2\".\n" }, { "name": "plugin, p", "content": "Specifies the plugin URI. You may need to escape ':'.\n" }, { "name": "controls, c", "content": "Set the '|' separated list of controls which are zero or more floating point values that\ndetermine the behavior of the loaded plugin (for example delay, threshold or gain). If\ncontrols is set to \"help\", all available controls and their valid ranges are printed.\n\nsamplerate, s\nSpecify the sample rate, default to 44100. Only used if plugin have zero inputs.\n\nnbsamples, n\nSet the number of samples per channel per each output frame, default is 1024. Only used\nif plugin have zero inputs.\n" }, { "name": "duration, d", "content": "Set the minimum duration of the sourced audio. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be\ngreater than the specified duration, as the generated audio is always cut at the end of a\ncomplete frame. If not specified, or the expressed duration is negative, the audio is\nsupposed to be generated forever. Only used if plugin have zero inputs.\n\nExamples\n\n• Apply bass enhancer plugin from Calf:\n\nlv2=p=http\\\\\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2\n\n• Apply vinyl plugin from Calf:\n\nlv2=p=http\\\\\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5\n\n• Apply bit crusher plugin from ArtyFX:\n\nlv2=p=http\\\\\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3\n" }, { "name": "mcompand", "content": "Multiband Compress or expand the audio's dynamic range.\n\nThe input audio is divided into bands using 4th order Linkwitz-Riley IIRs. This is akin to\nthe crossover of a loudspeaker, and results in flat frequency response when absent compander\naction.\n\nIt accepts the following parameters:\n" }, { "name": "args", "content": "This option syntax is: attack,decay,[attack,decay..] soft-knee points crossoverfrequency\n[delay [initialvolume [gain]]] | attack,decay ... For explanation of each item refer to\ncompand filter documentation.\n" }, { "name": "pan", "content": "Mix channels with specific gain levels. The filter accepts the output channel layout followed\nby a set of channels definitions.\n\nThis filter is also designed to efficiently remap the channels of an audio stream.\n\nThe filter accepts parameters of the form: \"l|outdef|outdef|...\"\n\nl output channel layout or number of channels\n" }, { "name": "outdef", "content": "output channel specification, of the form:\n\"outname=[gain*]inname[(+-)[gain*]inname...]\"\n\noutname\noutput channel to define, either a channel name (FL, FR, etc.) or a channel number (c0,\nc1, etc.)\n" }, { "name": "gain", "content": "multiplicative coefficient for the channel, 1 leaving the volume unchanged\n\ninname\ninput channel to use, see outname for details; it is not possible to mix named and\nnumbered input channels\n\nIf the `=' in a channel specification is replaced by `<', then the gains for that\nspecification will be renormalized so that the total is 1, thus avoiding clipping noise.\n\nMixing examples\n\nFor example, if you want to down-mix from stereo to mono, but with a bigger factor for the\nleft channel:\n\npan=1c|c0=0.9*c0+0.1*c1\n\nA customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels\nsurround:\n\npan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR\n\nNote that ffmpeg integrates a default down-mix (and up-mix) system that should be preferred\n(see \"-ac\" option) unless you have very specific needs.\n\nRemapping examples\n\nThe channel remapping will be effective if, and only if:\n\n*\n*\n\nIf all these conditions are satisfied, the filter will notify the user (\"Pure channel mapping\ndetected\"), and use an optimized and lossless method to do the remapping.\n\nFor example, if you have a 5.1 source and want a stereo audio stream by dropping the extra\nchannels:\n\npan=\"stereo| c0=FL | c1=FR\"\n\nGiven the same source, you can also switch front left and front right channels and keep the\ninput channel layout:\n\npan=\"5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5\"\n\nIf the input is a stereo audio stream, you can mute the front left channel (and still keep\nthe stereo channel layout) with:\n\npan=\"stereo|c1=c1\"\n\nStill with a stereo audio stream input, you can copy the right channel in both front left and\nright:\n\npan=\"stereo| c0=FR | c1=FR\"\n" }, { "name": "replaygain", "content": "ReplayGain scanner filter. This filter takes an audio stream as an input and outputs it\nunchanged. At end of filtering it displays \"trackgain\" and \"trackpeak\".\n" }, { "name": "resample", "content": "Convert the audio sample format, sample rate and channel layout. It is not meant to be used\ndirectly.\n" }, { "name": "rubberband", "content": "Apply time-stretching and pitch-shifting with librubberband.\n\nTo enable compilation of this filter, you need to configure FFmpeg with\n\"--enable-librubberband\".\n\nThe filter accepts the following options:\n" }, { "name": "tempo", "content": "Set tempo scale factor.\n" }, { "name": "pitch", "content": "Set pitch scale factor.\n" }, { "name": "transients", "content": "Set transients detector. Possible values are:\n\ncrisp\nmixed\nsmooth" }, { "name": "detector", "content": "Set detector. Possible values are:\n\ncompound\npercussive\nsoft" }, { "name": "phase", "content": "Set phase. Possible values are:\n\nlaminar\nindependent" }, { "name": "window", "content": "Set processing window size. Possible values are:\n\nstandard\nshort\nlong" }, { "name": "smoothing", "content": "Set smoothing. Possible values are:\n\noff\non" }, { "name": "formant", "content": "Enable formant preservation when shift pitching. Possible values are:\n\nshifted\npreserved" }, { "name": "pitchq", "content": "Set pitch quality. Possible values are:\n\nquality\nspeed\nconsistency" }, { "name": "channels", "content": "Set channels. Possible values are:\n\napart\ntogether\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "tempo", "content": "Change filter tempo scale factor. Syntax for the command is : \"tempo\"\n" }, { "name": "pitch", "content": "Change filter pitch scale factor. Syntax for the command is : \"pitch\"\n" }, { "name": "sidechaincompress", "content": "This filter acts like normal compressor but has the ability to compress detected signal using\nsecond input signal. It needs two input streams and returns one output stream. First input\nstream will be processed depending on second stream signal. The filtered signal then can be\nfiltered with other filters in later stages of processing. See pan and amerge filter.\n\nThe filter accepts the following options:\n\nlevelin\nSet input gain. Default is 1. Range is between 0.015625 and 64.\n" }, { "name": "mode", "content": "Set mode of compressor operation. Can be \"upward\" or \"downward\". Default is \"downward\".\n" }, { "name": "threshold", "content": "If a signal of second stream raises above this level it will affect the gain reduction of\nfirst stream. By default is 0.125. Range is between 0.00097563 and 1.\n" }, { "name": "ratio", "content": "Set a ratio about which the signal is reduced. 1:2 means that if the level raised 4dB\nabove the threshold, it will be only 2dB above after the reduction. Default is 2. Range\nis between 1 and 20.\n" }, { "name": "attack", "content": "Amount of milliseconds the signal has to rise above the threshold before gain reduction\nstarts. Default is 20. Range is between 0.01 and 2000.\n" }, { "name": "release", "content": "Amount of milliseconds the signal has to fall below the threshold before reduction is\ndecreased again. Default is 250. Range is between 0.01 and 9000.\n" }, { "name": "makeup", "content": "Set the amount by how much signal will be amplified after processing. Default is 1.\nRange is from 1 to 64.\n" }, { "name": "knee", "content": "Curve the sharp knee around the threshold to enter gain reduction more softly. Default\nis 2.82843. Range is between 1 and 8.\n" }, { "name": "link", "content": "Choose if the \"average\" level between all channels of side-chain stream or the\nlouder(\"maximum\") channel of side-chain stream affects the reduction. Default is\n\"average\".\n" }, { "name": "detection", "content": "Should the exact signal be taken in case of \"peak\" or an RMS one in case of \"rms\".\nDefault is \"rms\" which is mainly smoother.\n\nlevelsc\nSet sidechain gain. Default is 1. Range is between 0.015625 and 64.\n\nmix How much to use compressed signal in output. Default is 1. Range is between 0 and 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n\nExamples\n\n• Full ffmpeg example taking 2 audio inputs, 1st input to be compressed depending on the\nsignal of 2nd input and later compressed signal to be merged with 2nd input:\n\nffmpeg -i main.flac -i sidechain.flac -filtercomplex \"[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge\"\n" }, { "name": "sidechaingate", "content": "A sidechain gate acts like a normal (wideband) gate but has the ability to filter the\ndetected signal before sending it to the gain reduction stage. Normally a gate uses the full\nrange signal to detect a level above the threshold. For example: If you cut all lower\nfrequencies from your sidechain signal the gate will decrease the volume of your track only\nif not enough highs appear. With this technique you are able to reduce the resonation of a\nnatural drum or remove \"rumbling\" of muted strokes from a heavily distorted guitar. It needs\ntwo input streams and returns one output stream. First input stream will be processed\ndepending on second stream signal.\n\nThe filter accepts the following options:\n\nlevelin\nSet input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.\n" }, { "name": "mode", "content": "Set the mode of operation. Can be \"upward\" or \"downward\". Default is \"downward\". If set\nto \"upward\" mode, higher parts of signal will be amplified, expanding dynamic range in\nupward direction. Otherwise, in case of \"downward\" lower parts of signal will be\nreduced.\n" }, { "name": "range", "content": "Set the level of gain reduction when the signal is below the threshold. Default is\n0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then\nfilter behaves like expander.\n" }, { "name": "threshold", "content": "If a signal rises above this level the gain reduction is released. Default is 0.125.\nAllowed range is from 0 to 1.\n" }, { "name": "ratio", "content": "Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to\n9000.\n" }, { "name": "attack", "content": "Amount of milliseconds the signal has to rise above the threshold before gain reduction\nstops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.\n" }, { "name": "release", "content": "Amount of milliseconds the signal has to fall below the threshold before the reduction is\nincreased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.\n" }, { "name": "makeup", "content": "Set amount of amplification of signal after processing. Default is 1. Allowed range is\nfrom 1 to 64.\n" }, { "name": "knee", "content": "Curve the sharp knee around the threshold to enter gain reduction more softly. Default\nis 2.828427125. Allowed range is from 1 to 8.\n" }, { "name": "detection", "content": "Choose if exact signal should be taken for detection or an RMS like one. Default is rms.\nCan be peak or rms.\n" }, { "name": "link", "content": "Choose if the average level between all channels or the louder channel affects the\nreduction. Default is average. Can be average or maximum.\n\nlevelsc\nSet sidechain gain. Default is 1. Range is from 0.015625 to 64.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "silencedetect", "content": "Detect silence in an audio stream.\n\nThis filter logs a message when it detects that the input audio volume is less or equal to a\nnoise tolerance value for a duration greater or equal to the minimum detected noise duration.\n\nThe printed times and duration are expressed in seconds. The \"lavfi.silencestart\" or\n\"lavfi.silencestart.X\" metadata key is set on the first frame whose timestamp equals or\nexceeds the detection duration and it contains the timestamp of the first frame of the\nsilence.\n\nThe \"lavfi.silenceduration\" or \"lavfi.silenceduration.X\" and \"lavfi.silenceend\" or\n\"lavfi.silenceend.X\" metadata keys are set on the first frame after the silence. If mono is\nenabled, and each channel is evaluated separately, the \".X\" suffixed keys are used, and \"X\"\ncorresponds to the channel number.\n\nThe filter accepts the following options:\n" }, { "name": "noise, n", "content": "Set noise tolerance. Can be specified in dB (in case \"dB\" is appended to the specified\nvalue) or amplitude ratio. Default is -60dB, or 0.001.\n" }, { "name": "duration, d", "content": "Set silence duration until notification (default is 2 seconds). See the Time duration\nsection in the ffmpeg-utils(1) manual for the accepted syntax.\n" }, { "name": "mono, m", "content": "Process each channel separately, instead of combined. By default is disabled.\n\nExamples\n\n• Detect 5 seconds of silence with -50dB noise tolerance:\n\nsilencedetect=n=-50dB:d=5\n\n• Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in\nsilence.mp3:\n\nffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -\n" }, { "name": "silenceremove", "content": "Remove silence from the beginning, middle or end of the audio.\n\nThe filter accepts the following options:\n\nstartperiods\nThis value is used to indicate if audio should be trimmed at beginning of the audio. A\nvalue of zero indicates no silence should be trimmed from the beginning. When specifying\na non-zero value, it trims audio up until it finds non-silence. Normally, when trimming\nsilence from beginning of audio the startperiods will be 1 but it can be increased to\nhigher values to trim all audio up to specific count of non-silence periods. Default\nvalue is 0.\n\nstartduration\nSpecify the amount of time that non-silence must be detected before it stops trimming\naudio. By increasing the duration, bursts of noises can be treated as silence and trimmed\noff. Default value is 0.\n\nstartthreshold\nThis indicates what sample value should be treated as silence. For digital audio, a value\nof 0 may be fine but for audio recorded from analog, you may wish to increase the value\nto account for background noise. Can be specified in dB (in case \"dB\" is appended to the\nspecified value) or amplitude ratio. Default value is 0.\n\nstartsilence\nSpecify max duration of silence at beginning that will be kept after trimming. Default is\n0, which is equal to trimming all samples detected as silence.\n\nstartmode\nSpecify mode of detection of silence end in start of multi-channel audio. Can be any or\nall. Default is any. With any, any sample that is detected as non-silence will cause\nstopped trimming of silence. With all, only if all channels are detected as non-silence\nwill cause stopped trimming of silence.\n\nstopperiods\nSet the count for trimming silence from the end of audio. To remove silence from the\nmiddle of a file, specify a stopperiods that is negative. This value is then treated as\na positive value and is used to indicate the effect should restart processing as\nspecified by startperiods, making it suitable for removing periods of silence in the\nmiddle of the audio. Default value is 0.\n\nstopduration\nSpecify a duration of silence that must exist before audio is not copied any more. By\nspecifying a higher duration, silence that is wanted can be left in the audio. Default\nvalue is 0.\n\nstopthreshold\nThis is the same as startthreshold but for trimming silence from the end of audio. Can\nbe specified in dB (in case \"dB\" is appended to the specified value) or amplitude ratio.\nDefault value is 0.\n\nstopsilence\nSpecify max duration of silence at end that will be kept after trimming. Default is 0,\nwhich is equal to trimming all samples detected as silence.\n\nstopmode\nSpecify mode of detection of silence start in end of multi-channel audio. Can be any or\nall. Default is any. With any, any sample that is detected as non-silence will cause\nstopped trimming of silence. With all, only if all channels are detected as non-silence\nwill cause stopped trimming of silence.\n" }, { "name": "detection", "content": "Set how is silence detected. Can be \"rms\" or \"peak\". Second is faster and works better\nwith digital silence which is exactly 0. Default value is \"rms\".\n" }, { "name": "window", "content": "Set duration in number of seconds used to calculate size of window in number of samples\nfor detecting silence. Default value is 0.02. Allowed range is from 0 to 10.\n\nExamples\n\n• The following example shows how this filter can be used to start a recording that does\nnot contain the delay at the start which usually occurs between pressing the record\nbutton and the start of the performance:\n\nsilenceremove=startperiods=1:startduration=5:startthreshold=0.02\n\n• Trim all silence encountered from beginning to end where there is more than 1 second of\nsilence in audio:\n\nsilenceremove=stopperiods=-1:stopduration=1:stopthreshold=-90dB\n\n• Trim all digital silence samples, using peak detection, from beginning to end where there\nis more than 0 samples of digital silence in audio and digital silence is detected in all\nchannels at same positions in stream:\n\nsilenceremove=window=0:detection=peak:stopmode=all:startmode=all:stopperiods=-1:stopthreshold=0\n" }, { "name": "sofalizer", "content": "SOFAlizer uses head-related transfer functions (HRTFs) to create virtual loudspeakers around\nthe user for binaural listening via headphones (audio formats up to 9 channels supported).\nThe HRTFs are stored in SOFA files (see for a database).\nSOFAlizer is developed at the Acoustics Research Institute (ARI) of the Austrian Academy of\nSciences.\n\nTo enable compilation of this filter you need to configure FFmpeg with \"--enable-libmysofa\".\n\nThe filter accepts the following options:\n" }, { "name": "sofa", "content": "Set the SOFA file used for rendering.\n" }, { "name": "gain", "content": "Set gain applied to audio. Value is in dB. Default is 0.\n" }, { "name": "rotation", "content": "Set rotation of virtual loudspeakers in deg. Default is 0.\n" }, { "name": "elevation", "content": "Set elevation of virtual speakers in deg. Default is 0.\n" }, { "name": "radius", "content": "Set distance in meters between loudspeakers and the listener with near-field HRTFs.\nDefault is 1.\n" }, { "name": "type", "content": "Set processing type. Can be time or freq. time is processing audio in time domain which\nis slow. freq is processing audio in frequency domain which is fast. Default is freq.\n" }, { "name": "speakers", "content": "Set custom positions of virtual loudspeakers. Syntax for this option is: \n[| |...]. Each virtual loudspeaker is described with short\nchannel name following with azimuth and elevation in degrees. Each virtual loudspeaker\ndescription is separated by '|'. For example to override front left and front right\nchannel positions use: 'speakers=FL 45 15|FR 345 15'. Descriptions with unrecognised\nchannel names are ignored.\n" }, { "name": "lfegain", "content": "Set custom gain for LFE channels. Value is in dB. Default is 0.\n" }, { "name": "framesize", "content": "Set custom frame size in number of samples. Default is 1024. Allowed range is from 1024\nto 96000. Only used if option type is set to freq.\n" }, { "name": "normalize", "content": "Should all IRs be normalized upon importing SOFA file. By default is enabled.\n" }, { "name": "interpolate", "content": "Should nearest IRs be interpolated with neighbor IRs if exact position does not match. By\ndefault is disabled.\n" }, { "name": "minphase", "content": "Minphase all IRs upon loading of SOFA file. By default is disabled.\n" }, { "name": "anglestep", "content": "Set neighbor search angle step. Only used if option interpolate is enabled.\n" }, { "name": "radstep", "content": "Set neighbor search radius step. Only used if option interpolate is enabled.\n\nExamples\n\n• Using ClubFritz6 sofa file:\n\nsofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1\n\n• Using ClubFritz12 sofa file and bigger radius with small rotation:\n\nsofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5\n\n• Similar as above but with custom speaker positions for front left, front right, back left\nand back right and also with custom gain:\n\n\"sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28\"\n" }, { "name": "speechnorm", "content": "Speech Normalizer.\n\nThis filter expands or compresses each half-cycle of audio samples (local set of samples all\nabove or all below zero and between two nearest zero crossings) depending on threshold value,\nso audio reaches target peak value under conditions controlled by below options.\n\nThe filter accepts the following options:\n" }, { "name": "peak, p", "content": "Set the expansion target peak value. This specifies the highest allowed absolute\namplitude level for the normalized audio input. Default value is 0.95. Allowed range is\nfrom 0.0 to 1.0.\n" }, { "name": "expansion, e", "content": "Set the maximum expansion factor. Allowed range is from 1.0 to 50.0. Default value is\n2.0. This option controls maximum local half-cycle of samples expansion. The maximum\nexpansion would be such that local peak value reaches target peak value but never to\nsurpass it and that ratio between new and previous peak value does not surpass this\noption value.\n" }, { "name": "compression, c", "content": "Set the maximum compression factor. Allowed range is from 1.0 to 50.0. Default value is\n2.0. This option controls maximum local half-cycle of samples compression. This option\nis used only if threshold option is set to value greater than 0.0, then in such cases\nwhen local peak is lower or same as value set by threshold all samples belonging to that\npeak's half-cycle will be compressed by current compression factor.\n" }, { "name": "threshold, t", "content": "Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to 1.0. This\noption specifies which half-cycles of samples will be compressed and which will be\nexpanded. Any half-cycle samples with their local peak value below or same as this\noption value will be compressed by current compression factor, otherwise, if greater than\nthreshold value they will be expanded with expansion factor so that it could reach peak\ntarget value but never surpass it.\n" }, { "name": "raise, r", "content": "Set the expansion raising amount per each half-cycle of samples. Default value is 0.001.\nAllowed range is from 0.0 to 1.0. This controls how fast expansion factor is raised per\neach new half-cycle until it reaches expansion value. Setting this options too high may\nlead to distortions.\n" }, { "name": "fall, f", "content": "Set the compression raising amount per each half-cycle of samples. Default value is\n0.001. Allowed range is from 0.0 to 1.0. This controls how fast compression factor is\nraised per each new half-cycle until it reaches compression value.\n" }, { "name": "channels, h", "content": "Specify which channels to filter, by default all available channels are filtered.\n" }, { "name": "invert, i", "content": "Enable inverted filtering, by default is disabled. This inverts interpretation of\nthreshold option. When enabled any half-cycle of samples with their local peak value\nbelow or same as threshold option will be expanded otherwise it will be compressed.\n" }, { "name": "link, l", "content": "Link channels when calculating gain applied to each filtered channel sample, by default\nis disabled. When disabled each filtered channel gain calculation is independent,\notherwise when this option is enabled the minimum of all possible gains for each filtered\nchannel is used.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "stereotools", "content": "This filter has some handy utilities to manage stereo signals, for converting M/S stereo\nrecordings to L/R signal while having control over the parameters or spreading the stereo\nimage of master track.\n\nThe filter accepts the following options:\n\nlevelin\nSet input level before filtering for both channels. Defaults is 1. Allowed range is from\n0.015625 to 64.\n\nlevelout\nSet output level after filtering for both channels. Defaults is 1. Allowed range is from\n0.015625 to 64.\n\nbalancein\nSet input balance between both channels. Default is 0. Allowed range is from -1 to 1.\n\nbalanceout\nSet output balance between both channels. Default is 0. Allowed range is from -1 to 1.\n" }, { "name": "softclip", "content": "Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping.\nDisabled by default.\n" }, { "name": "mutel", "content": "Mute the left channel. Disabled by default.\n" }, { "name": "muter", "content": "Mute the right channel. Disabled by default.\n" }, { "name": "phasel", "content": "Change the phase of the left channel. Disabled by default.\n" }, { "name": "phaser", "content": "Change the phase of the right channel. Disabled by default.\n" }, { "name": "mode", "content": "Set stereo mode. Available values are:\n\nlr>lr\nLeft/Right to Left/Right, this is default.\n\nlr>ms\nLeft/Right to Mid/Side.\n\nms>lr\nMid/Side to Left/Right.\n\nlr>ll\nLeft/Right to Left/Left.\n\nlr>rr\nLeft/Right to Right/Right.\n\nlr>l+r\nLeft/Right to Left + Right.\n\nlr>rl\nLeft/Right to Right/Left.\n\nms>ll\nMid/Side to Left/Left.\n\nms>rr\nMid/Side to Right/Right.\n\nms>rl\nMid/Side to Right/Left.\n\nlr>l-r\nLeft/Right to Left - Right.\n" }, { "name": "slev", "content": "Set level of side signal. Default is 1. Allowed range is from 0.015625 to 64.\n" }, { "name": "sbal", "content": "Set balance of side signal. Default is 0. Allowed range is from -1 to 1.\n" }, { "name": "mlev", "content": "Set level of the middle signal. Default is 1. Allowed range is from 0.015625 to 64.\n" }, { "name": "mpan", "content": "Set middle signal pan. Default is 0. Allowed range is from -1 to 1.\n" }, { "name": "base", "content": "Set stereo base between mono and inversed channels. Default is 0. Allowed range is from\n-1 to 1.\n" }, { "name": "delay", "content": "Set delay in milliseconds how much to delay left from right channel and vice versa.\nDefault is 0. Allowed range is from -20 to 20.\n" }, { "name": "sclevel", "content": "Set S/C level. Default is 1. Allowed range is from 1 to 100.\n" }, { "name": "phase", "content": "Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.\n\nbmodein, bmodeout\nSet balance mode for balancein/balanceout option.\n\nCan be one of the following:\n\nbalance\nClassic balance mode. Attenuate one channel at time. Gain is raised up to 1.\n\namplitude\nSimilar as classic mode above but gain is raised up to 2.\n\npower\nEqual power distribution, from -6dB to +6dB range.\n\nCommands\n\nThis filter supports the all above options as commands.\n\nExamples\n\n• Apply karaoke like effect:\n\nstereotools=mlev=0.015625\n\n• Convert M/S signal to L/R:\n\n\"stereotools=mode=ms>lr\"\n" }, { "name": "stereowiden", "content": "This filter enhance the stereo effect by suppressing signal common to both channels and by\ndelaying the signal of left into right and vice versa, thereby widening the stereo effect.\n\nThe filter accepts the following options:\n" }, { "name": "delay", "content": "Time in milliseconds of the delay of left signal into right and vice versa. Default is\n20 milliseconds.\n" }, { "name": "feedback", "content": "Amount of gain in delayed signal into right and vice versa. Gives a delay effect of left\nsignal in right output and vice versa which gives widening effect. Default is 0.3.\n" }, { "name": "crossfeed", "content": "Cross feed of left into right with inverted phase. This helps in suppressing the mono. If\nthe value is 1 it will cancel all the signal common to both channels. Default is 0.3.\n" }, { "name": "drymix", "content": "Set level of input signal of original channel. Default is 0.8.\n\nCommands\n\nThis filter supports the all above options except \"delay\" as commands.\n" }, { "name": "superequalizer", "content": "Apply 18 band equalizer.\n\nThe filter accepts the following options:\n\n1b Set 65Hz band gain.\n\n2b Set 92Hz band gain.\n\n3b Set 131Hz band gain.\n\n4b Set 185Hz band gain.\n\n5b Set 262Hz band gain.\n\n6b Set 370Hz band gain.\n\n7b Set 523Hz band gain.\n\n8b Set 740Hz band gain.\n\n9b Set 1047Hz band gain.\n\n10b Set 1480Hz band gain.\n\n11b Set 2093Hz band gain.\n\n12b Set 2960Hz band gain.\n\n13b Set 4186Hz band gain.\n\n14b Set 5920Hz band gain.\n\n15b Set 8372Hz band gain.\n\n16b Set 11840Hz band gain.\n\n17b Set 16744Hz band gain.\n\n18b Set 20000Hz band gain.\n" }, { "name": "surround", "content": "Apply audio surround upmix filter.\n\nThis filter allows to produce multichannel output from audio stream.\n\nThe filter accepts the following options:\n\nchlout\nSet output channel layout. By default, this is 5.1.\n\nSee the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.\n\nchlin\nSet input channel layout. By default, this is stereo.\n\nSee the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.\n\nlevelin\nSet input volume level. By default, this is 1.\n\nlevelout\nSet output volume level. By default, this is 1.\n\nlfe Enable LFE channel output if output channel layout has it. By default, this is enabled.\n\nlfelow\nSet LFE low cut off frequency. By default, this is 128 Hz.\n\nlfehigh\nSet LFE high cut off frequency. By default, this is 256 Hz.\n\nlfemode\nSet LFE mode, can be add or sub. Default is add. In add mode, LFE channel is created\nfrom input audio and added to output. In sub mode, LFE channel is created from input\naudio and added to output but also all non-LFE output channels are subtracted with output\nLFE channel.\n" }, { "name": "angle", "content": "Set angle of stereo surround transform, Allowed range is from 0 to 360. Default is 90.\n\nfcin\nSet front center input volume. By default, this is 1.\n\nfcout\nSet front center output volume. By default, this is 1.\n\nflin\nSet front left input volume. By default, this is 1.\n\nflout\nSet front left output volume. By default, this is 1.\n\nfrin\nSet front right input volume. By default, this is 1.\n\nfrout\nSet front right output volume. By default, this is 1.\n\nslin\nSet side left input volume. By default, this is 1.\n\nslout\nSet side left output volume. By default, this is 1.\n\nsrin\nSet side right input volume. By default, this is 1.\n\nsrout\nSet side right output volume. By default, this is 1.\n\nblin\nSet back left input volume. By default, this is 1.\n\nblout\nSet back left output volume. By default, this is 1.\n\nbrin\nSet back right input volume. By default, this is 1.\n\nbrout\nSet back right output volume. By default, this is 1.\n\nbcin\nSet back center input volume. By default, this is 1.\n\nbcout\nSet back center output volume. By default, this is 1.\n\nlfein\nSet LFE input volume. By default, this is 1.\n\nlfeout\nSet LFE output volume. By default, this is 1.\n" }, { "name": "allx", "content": "Set spread usage of stereo image across X axis for all channels.\n" }, { "name": "ally", "content": "Set spread usage of stereo image across Y axis for all channels.\n" }, { "name": "fcx, flx, frx, blx, brx, slx, srx, bcx", "content": "Set spread usage of stereo image across X axis for each channel.\n" }, { "name": "fcy, fly, fry, bly, bry, sly, sry, bcy", "content": "Set spread usage of stereo image across Y axis for each channel.\n\nwinsize\nSet window size. Allowed range is from 1024 to 65536. Default size is 4096.\n\nwinfunc\nSet window function.\n\nIt accepts the following values:\n\nrect\nbartlett\nhann, hanning\nhamming\nblackman\nwelch\nflattop\nbharris\nbnuttall\nbhann\nsine\nnuttall\nlanczos\ngauss\ntukey\ndolph\ncauchy\nparzen\npoisson\nbohman\n\nDefault is \"hann\".\n" }, { "name": "overlap", "content": "Set window overlap. If set to 1, the recommended overlap for selected window function\nwill be picked. Default is 0.5.\n" }, { "name": "treble, highshelf", "content": "Boost or cut treble (upper) frequencies of the audio using a two-pole shelving filter with a\nresponse similar to that of a standard hi-fi's tone-controls. This is also known as shelving\nequalisation (EQ).\n\nThe filter accepts the following options:\n" }, { "name": "gain, g", "content": "Give the gain at whichever is the lower of ~22 kHz and the Nyquist frequency. Its useful\nrange is about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when\nusing a positive gain.\n" }, { "name": "frequency, f", "content": "Set the filter's central frequency and so can be used to extend or reduce the frequency\nrange to be boosted or cut. The default value is 3000 Hz.\n\nwidthtype, t\nSet method to specify band-width of filter.\n\nh Hz\n\nq Q-Factor\n\no octave\n\ns slope\n\nk kHz\n" }, { "name": "width, w", "content": "Determine how steep is the filter's shelf transition.\n" }, { "name": "poles, p", "content": "Set number of poles. Default is 2.\n" }, { "name": "mix, m", "content": "How much to use filtered signal in output. Default is 1. Range is between 0 and 1.\n" }, { "name": "channels, c", "content": "Specify which channels to filter, by default all available are filtered.\n" }, { "name": "normalize, n", "content": "Normalize biquad coefficients, by default is disabled. Enabling it will normalize\nmagnitude response at DC to 0dB.\n" }, { "name": "transform, a", "content": "Set transform type of IIR filter.\n\ndi\ndii\ntdii\nlatt" }, { "name": "precision, r", "content": "Set precison of filtering.\n\nauto\nPick automatic sample format depending on surround filters.\n\ns16 Always use signed 16-bit.\n\ns32 Always use signed 32-bit.\n\nf32 Always use float 32-bit.\n\nf64 Always use float 64-bit.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "frequency, f", "content": "Change treble frequency. Syntax for the command is : \"frequency\"\n\nwidthtype, t\nChange treble widthtype. Syntax for the command is : \"widthtype\"\n" }, { "name": "width, w", "content": "Change treble width. Syntax for the command is : \"width\"\n" }, { "name": "gain, g", "content": "Change treble gain. Syntax for the command is : \"gain\"\n" }, { "name": "mix, m", "content": "Change treble mix. Syntax for the command is : \"mix\"\n" }, { "name": "tremolo", "content": "Sinusoidal amplitude modulation.\n\nThe filter accepts the following options:\n\nf Modulation frequency in Hertz. Modulation frequencies in the subharmonic range (20 Hz or\nlower) will result in a tremolo effect. This filter may also be used as a ring modulator\nby specifying a modulation frequency higher than 20 Hz. Range is 0.1 - 20000.0. Default\nvalue is 5.0 Hz.\n\nd Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.\n" }, { "name": "vibrato", "content": "Sinusoidal phase modulation.\n\nThe filter accepts the following options:\n\nf Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.\n\nd Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.\n" }, { "name": "volume", "content": "Adjust the input audio volume.\n\nIt accepts the following parameters:\n" }, { "name": "volume", "content": "Set audio volume expression.\n\nOutput values are clipped to the maximum value.\n\nThe output audio volume is given by the relation:\n\n = * \n\nThe default value for volume is \"1.0\".\n" }, { "name": "precision", "content": "This parameter represents the mathematical precision.\n\nIt determines which input sample formats will be allowed, which affects the precision of\nthe volume scaling.\n\nfixed\n8-bit fixed-point; this limits input sample format to U8, S16, and S32.\n\nfloat\n32-bit floating-point; this limits input sample format to FLT. (default)\n\ndouble\n64-bit floating-point; this limits input sample format to DBL.\n" }, { "name": "replaygain", "content": "Choose the behaviour on encountering ReplayGain side data in input frames.\n\ndrop\nRemove ReplayGain side data, ignoring its contents (the default).\n\nignore\nIgnore ReplayGain side data, but leave it in the frame.\n\ntrack\nPrefer the track gain, if present.\n\nalbum\nPrefer the album gain, if present.\n\nreplaygainpreamp\nPre-amplification gain in dB to apply to the selected replaygain gain.\n\nDefault value for replaygainpreamp is 0.0.\n\nreplaygainnoclip\nPrevent clipping by limiting the gain applied.\n\nDefault value for replaygainnoclip is 1.\n" }, { "name": "eval", "content": "Set when the volume expression is evaluated.\n\nIt accepts the following values:\n\nonce\nonly evaluate expression once during the filter initialization, or when the volume\ncommand is sent\n\nframe\nevaluate expression for each incoming frame\n\nDefault value is once.\n\nThe volume expression can contain the following parameters.\n\nn frame number (starting at zero)\n\nnbchannels\nnumber of channels\n\nnbconsumedsamples\nnumber of samples consumed by the filter\n\nnbsamples\nnumber of samples in the current frame\n\npos original frame position in the file\n\npts frame PTS\n\nsamplerate\nsample rate\n" }, { "name": "startpts", "content": "PTS at start of stream\n" }, { "name": "startt", "content": "time at start of stream\n\nt frame time\n\ntb timestamp timebase\n" }, { "name": "volume", "content": "last set volume value\n\nNote that when eval is set to once only the samplerate and tb variables are available, all\nother variables will evaluate to NAN.\n\nCommands\n\nThis filter supports the following commands:\n" }, { "name": "volume", "content": "Modify the volume expression. The command accepts the same syntax of the corresponding\noption.\n\nIf the specified expression is not valid, it is kept at its current value.\n\nExamples\n\n• Halve the input audio volume:\n\nvolume=volume=0.5\nvolume=volume=1/2\nvolume=volume=-6.0206dB\n\nIn all the above example the named key for volume can be omitted, for example like in:\n\nvolume=0.5\n\n• Increase input audio power by 6 decibels using fixed-point precision:\n\nvolume=volume=6dB:precision=fixed\n\n• Fade volume after time 10 with an annihilation period of 5 seconds:\n\nvolume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame\n" }, { "name": "volumedetect", "content": "Detect the volume of the input video.\n\nThe filter has no parameters. The input is not modified. Statistics about the volume will be\nprinted in the log when the input stream end is reached.\n\nIn particular it will show the mean volume (root mean square), maximum volume (on a per-\nsample basis), and the beginning of a histogram of the registered volume values (from the\nmaximum value to a cumulated 1/1000 of the samples).\n\nAll volumes are in decibels relative to the maximum PCM value.\n\nExamples\n\nHere is an excerpt of the output:\n\n[Parsedvolumedetect0 0xa23120] meanvolume: -27 dB\n[Parsedvolumedetect0 0xa23120] maxvolume: -4 dB\n[Parsedvolumedetect0 0xa23120] histogram4db: 6\n[Parsedvolumedetect0 0xa23120] histogram5db: 62\n[Parsedvolumedetect0 0xa23120] histogram6db: 286\n[Parsedvolumedetect0 0xa23120] histogram7db: 1042\n[Parsedvolumedetect0 0xa23120] histogram8db: 2551\n[Parsedvolumedetect0 0xa23120] histogram9db: 4609\n[Parsedvolumedetect0 0xa23120] histogram10db: 8409\n\nIt means that:\n\n• The mean square energy is approximately -27 dB, or 10^-2.7.\n\n• The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.\n\n• There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.\n\nIn other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB\ncauses clipping for 6 samples, etc.\n" } ] }, "AUDIO SOURCES": { "content": "Below is a description of the currently available audio sources.\n", "subsections": [ { "name": "abuffer", "content": "Buffer audio frames, and make them available to the filter chain.\n\nThis source is mainly intended for a programmatic use, in particular through the interface\ndefined in libavfilter/buffersrc.h.\n\nIt accepts the following parameters:\n\ntimebase\nThe timebase which will be used for timestamps of submitted frames. It must be either a\nfloating-point number or in numerator/denominator form.\n\nsamplerate\nThe sample rate of the incoming audio buffers.\n\nsamplefmt\nThe sample format of the incoming audio buffers. Either a sample format name or its\ncorresponding integer representation from the enum AVSampleFormat in\nlibavutil/samplefmt.h\n\nchannellayout\nThe channel layout of the incoming audio buffers. Either a channel layout name from\nchannellayoutmap in libavutil/channellayout.c or its corresponding integer\nrepresentation from the AVCHLAYOUT* macros in libavutil/channellayout.h\n" }, { "name": "channels", "content": "The number of channels of the incoming audio buffers. If both channels and\nchannellayout are specified, then they must be consistent.\n\nExamples\n\nabuffer=samplerate=44100:samplefmt=s16p:channellayout=stereo\n\nwill instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample\nformat with name \"s16p\" corresponds to the number 6 and the \"stereo\" channel layout\ncorresponds to the value 0x3, this is equivalent to:\n\nabuffer=samplerate=44100:samplefmt=6:channellayout=0x3\n" }, { "name": "aevalsrc", "content": "Generate an audio signal specified by an expression.\n\nThis source accepts in input one or more expressions (one for each channel), which are\nevaluated and used to generate a corresponding audio signal.\n\nThis source accepts the following options:\n" }, { "name": "exprs", "content": "Set the '|'-separated expressions list for each separate channel. In case the\nchannellayout option is not specified, the selected channel layout depends on the number\nof provided expressions. Otherwise the last specified expression is applied to the\nremaining output channels.\n\nchannellayout, c\nSet the channel layout. The number of channels in the specified layout must be equal to\nthe number of specified expressions.\n" }, { "name": "duration, d", "content": "Set the minimum duration of the sourced audio. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be\ngreater than the specified duration, as the generated audio is always cut at the end of a\ncomplete frame.\n\nIf not specified, or the expressed duration is negative, the audio is supposed to be\ngenerated forever.\n\nnbsamples, n\nSet the number of samples per channel per each output frame, default to 1024.\n\nsamplerate, s\nSpecify the sample rate, default to 44100.\n\nEach expression in exprs can contain the following constants:\n\nn number of the evaluated sample, starting from 0\n\nt time of the evaluated sample expressed in seconds, starting from 0\n\ns sample rate\n\nExamples\n\n• Generate silence:\n\naevalsrc=0\n\n• Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:\n\naevalsrc=\"sin(440*2*PI*t):s=8000\"\n\n• Generate a two channels signal, specify the channel layout (Front Center + Back Center)\nexplicitly:\n\naevalsrc=\"sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC\"\n\n• Generate white noise:\n\naevalsrc=\"-2+random(0)\"\n\n• Generate an amplitude modulated signal:\n\naevalsrc=\"sin(10*2*PI*t)*sin(880*2*PI*t)\"\n\n• Generate 2.5 Hz binaural beats on a 360 Hz carrier:\n\naevalsrc=\"0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)\"\n" }, { "name": "afirsrc", "content": "Generate a FIR coefficients using frequency sampling method.\n\nThe resulting stream can be used with afir filter for filtering the audio signal.\n\nThe filter accepts the following options:\n" }, { "name": "taps, t", "content": "Set number of filter coefficents in output audio stream. Default value is 1025.\n" }, { "name": "frequency, f", "content": "Set frequency points from where magnitude and phase are set. This must be in non\ndecreasing order, and first element must be 0, while last element must be 1. Elements are\nseparated by white spaces.\n" }, { "name": "magnitude, m", "content": "Set magnitude value for every frequency point set by frequency. Number of values must be\nsame as number of frequency points. Values are separated by white spaces.\n" }, { "name": "phase, p", "content": "Set phase value for every frequency point set by frequency. Number of values must be\nsame as number of frequency points. Values are separated by white spaces.\n\nsamplerate, r\nSet sample rate, default is 44100.\n\nnbsamples, n\nSet number of samples per each frame. Default is 1024.\n\nwinfunc, w\nSet window function. Default is blackman.\n" }, { "name": "anullsrc", "content": "The null audio source, return unprocessed audio frames. It is mainly useful as a template and\nto be employed in analysis / debugging tools, or as the source for filters which ignore the\ninput data (for example the sox synth filter).\n\nThis source accepts the following options:\n\nchannellayout, cl\nSpecifies the channel layout, and can be either an integer or a string representing a\nchannel layout. The default value of channellayout is \"stereo\".\n\nCheck the channellayoutmap definition in libavutil/channellayout.c for the mapping\nbetween strings and channel layout values.\n\nsamplerate, r\nSpecifies the sample rate, and defaults to 44100.\n\nnbsamples, n\nSet the number of samples per requested frames.\n" }, { "name": "duration, d", "content": "Set the duration of the sourced audio. See the Time duration section in the\nffmpeg-utils(1) manual for the accepted syntax.\n\nIf not specified, or the expressed duration is negative, the audio is supposed to be\ngenerated forever.\n\nExamples\n\n• Set the sample rate to 48000 Hz and the channel layout to AVCHLAYOUTMONO.\n\nanullsrc=r=48000:cl=4\n\n• Do the same operation with a more obvious syntax:\n\nanullsrc=r=48000:cl=mono\n\nAll the parameters need to be explicitly defined.\n" }, { "name": "flite", "content": "Synthesize a voice utterance using the libflite library.\n\nTo enable compilation of this filter you need to configure FFmpeg with \"--enable-libflite\".\n\nNote that versions of the flite library prior to 2.0 are not thread-safe.\n\nThe filter accepts the following options:\n\nlistvoices\nIf set to 1, list the names of the available voices and exit immediately. Default value\nis 0.\n\nnbsamples, n\nSet the maximum number of samples per frame. Default value is 512.\n" }, { "name": "textfile", "content": "Set the filename containing the text to speak.\n" }, { "name": "text", "content": "Set the text to speak.\n" }, { "name": "voice, v", "content": "Set the voice to use for the speech synthesis. Default value is \"kal\". See also the\nlistvoices option.\n\nExamples\n\n• Read from file speech.txt, and synthesize the text using the standard flite voice:\n\nflite=textfile=speech.txt\n\n• Read the specified text selecting the \"slt\" voice:\n\nflite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt\n\n• Input text to ffmpeg:\n\nffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt\n\n• Make ffplay speak the specified text, using \"flite\" and the \"lavfi\" device:\n\nffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'\n\nFor more information about libflite, check: \n" }, { "name": "anoisesrc", "content": "Generate a noise audio signal.\n\nThe filter accepts the following options:\n\nsamplerate, r\nSpecify the sample rate. Default value is 48000 Hz.\n" }, { "name": "amplitude, a", "content": "Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.\n" }, { "name": "duration, d", "content": "Specify the duration of the generated audio stream. Not specifying this option results in\nnoise with an infinite length.\n" }, { "name": "color, colour, c", "content": "Specify the color of noise. Available noise colors are white, pink, brown, blue, violet\nand velvet. Default color is white.\n" }, { "name": "seed, s", "content": "Specify a value used to seed the PRNG.\n\nnbsamples, n\nSet the number of samples per each output frame, default is 1024.\n\nExamples\n\n• Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:\n\nanoisesrc=d=60:c=pink:r=44100:a=0.5\n" }, { "name": "hilbert", "content": "Generate odd-tap Hilbert transform FIR coefficients.\n\nThe resulting stream can be used with afir filter for phase-shifting the signal by 90\ndegrees.\n\nThis is used in many matrix coding schemes and for analytic signal generation. The process\nis often written as a multiplication by i (or j), the imaginary unit.\n\nThe filter accepts the following options:\n\nsamplerate, s\nSet sample rate, default is 44100.\n" }, { "name": "taps, t", "content": "Set length of FIR filter, default is 22051.\n\nnbsamples, n\nSet number of samples per each frame.\n\nwinfunc, w\nSet window function to be used when generating FIR coefficients.\n" }, { "name": "sinc", "content": "Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR\ncoefficients.\n\nThe resulting stream can be used with afir filter for filtering the audio signal.\n\nThe filter accepts the following options:\n\nsamplerate, r\nSet sample rate, default is 44100.\n\nnbsamples, n\nSet number of samples per each frame. Default is 1024.\n\nhp Set high-pass frequency. Default is 0.\n\nlp Set low-pass frequency. Default is 0. If high-pass frequency is lower than low-pass\nfrequency and low-pass frequency is higher than 0 then filter will create band-pass\nfilter coefficients, otherwise band-reject filter coefficients.\n" }, { "name": "phase", "content": "Set filter phase response. Default is 50. Allowed range is from 0 to 100.\n" }, { "name": "beta", "content": "Set Kaiser window beta.\n\natt Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB.\n" }, { "name": "round", "content": "Enable rounding, by default is disabled.\n" }, { "name": "hptaps", "content": "Set number of taps for high-pass filter.\n" }, { "name": "lptaps", "content": "Set number of taps for low-pass filter.\n" }, { "name": "sine", "content": "Generate an audio signal made of a sine wave with amplitude 1/8.\n\nThe audio signal is bit-exact.\n\nThe filter accepts the following options:\n" }, { "name": "frequency, f", "content": "Set the carrier frequency. Default is 440 Hz.\n\nbeepfactor, b\nEnable a periodic beep every second with frequency beepfactor times the carrier\nfrequency. Default is 0, meaning the beep is disabled.\n\nsamplerate, r\nSpecify the sample rate, default is 44100.\n" }, { "name": "duration, d", "content": "Specify the duration of the generated audio stream.\n\nsamplesperframe\nSet the number of samples per output frame.\n\nThe expression can contain the following constants:\n\nn The (sequential) number of the output audio frame, starting from 0.\n\npts The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.\n\nt The PTS of the output audio frame, expressed in seconds.\n\nTB The timebase of the output audio frames.\n\nDefault is 1024.\n\nExamples\n\n• Generate a simple 440 Hz sine wave:\n\nsine\n\n• Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:\n\nsine=220:4:d=5\nsine=f=220:b=4:d=5\nsine=frequency=220:beepfactor=4:duration=5\n\n• Generate a 1 kHz sine wave following \"1602,1601,1602,1601,1602\" NTSC pattern:\n\nsine=1000:samplesperframe='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'\n" } ] }, "AUDIO SINKS": { "content": "Below is a description of the currently available audio sinks.\n", "subsections": [ { "name": "abuffersink", "content": "Buffer audio frames, and make them available to the end of filter chain.\n\nThis sink is mainly intended for programmatic use, in particular through the interface\ndefined in libavfilter/buffersink.h or the options system.\n\nIt accepts a pointer to an AVABufferSinkContext structure, which defines the incoming\nbuffers' formats, to be passed as the opaque parameter to \"avfilterinitfilter\" for\ninitialization.\n" }, { "name": "anullsink", "content": "Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a\ntemplate and for use in analysis / debugging tools.\n" } ] }, "VIDEO FILTERS": { "content": "When you configure your FFmpeg build, you can disable any of the existing filters using\n\"--disable-filters\". The configure output will show the video filters included in your\nbuild.\n\nBelow is a description of the currently available video filters.\n", "subsections": [ { "name": "addroi", "content": "Mark a region of interest in a video frame.\n\nThe frame data is passed through unchanged, but metadata is attached to the frame indicating\nregions of interest which can affect the behaviour of later encoding. Multiple regions can\nbe marked by applying the filter multiple times.\n\nx Region distance in pixels from the left edge of the frame.\n\ny Region distance in pixels from the top edge of the frame.\n\nw Region width in pixels.\n\nh Region height in pixels.\n\nThe parameters x, y, w and h are expressions, and may contain the following variables:\n\niw Width of the input frame.\n\nih Height of the input frame.\n" }, { "name": "qoffset", "content": "Quantisation offset to apply within the region.\n\nThis must be a real value in the range -1 to +1. A value of zero indicates no quality\nchange. A negative value asks for better quality (less quantisation), while a positive\nvalue asks for worse quality (greater quantisation).\n\nThe range is calibrated so that the extreme values indicate the largest possible offset -\nif the rest of the frame is encoded with the worst possible quality, an offset of -1\nindicates that this region should be encoded with the best possible quality anyway.\nIntermediate values are then interpolated in some codec-dependent way.\n\nFor example, in 10-bit H.264 the quantisation parameter varies between -12 and 51. A\ntypical qoffset value of -1/10 therefore indicates that this region should be encoded\nwith a QP around one-tenth of the full range better than the rest of the frame. So, if\nmost of the frame were to be encoded with a QP of around 30, this region would get a QP\nof around 24 (an offset of approximately -1/10 * (51 - -12) = -6.3). An extreme value of\n-1 would indicate that this region should be encoded with the best possible quality\nregardless of the treatment of the rest of the frame - that is, should be encoded at a QP\nof -12.\n" }, { "name": "clear", "content": "If set to true, remove any existing regions of interest marked on the frame before adding\nthe new one.\n\nExamples\n\n• Mark the centre quarter of the frame as interesting.\n\naddroi=iw/4:ih/4:iw/2:ih/2:-1/10\n\n• Mark the 100-pixel-wide region on the left edge of the frame as very uninteresting (to be\nencoded at much lower quality than the rest of the frame).\n\naddroi=0:0:100:ih:+1/5\n" }, { "name": "alphaextract", "content": "Extract the alpha component from the input as a grayscale video. This is especially useful\nwith the alphamerge filter.\n" }, { "name": "alphamerge", "content": "Add or replace the alpha component of the primary input with the grayscale value of a second\ninput. This is intended for use with alphaextract to allow the transmission or storage of\nframe sequences that have alpha in a format that doesn't support an alpha channel.\n\nFor example, to reconstruct full frames from a normal YUV-encoded video and a separate video\ncreated with alphaextract, you might use:\n\nmovie=inalpha.mkv [alpha]; [in][alpha] alphamerge [out]\n" }, { "name": "amplify", "content": "Amplify differences between current pixel and pixels of adjacent frames in same pixel\nlocation.\n\nThis filter accepts the following options:\n" }, { "name": "radius", "content": "Set frame radius. Default is 2. Allowed range is from 1 to 63. For example radius of 3\nwill instruct filter to calculate average of 7 frames.\n" }, { "name": "factor", "content": "Set factor to amplify difference. Default is 2. Allowed range is from 0 to 65535.\n" }, { "name": "threshold", "content": "Set threshold for difference amplification. Any difference greater or equal to this value\nwill not alter source pixel. Default is 10. Allowed range is from 0 to 65535.\n" }, { "name": "tolerance", "content": "Set tolerance for difference amplification. Any difference lower to this value will not\nalter source pixel. Default is 0. Allowed range is from 0 to 65535.\n\nlow Set lower limit for changing source pixel. Default is 65535. Allowed range is from 0 to\n65535. This option controls maximum possible value that will decrease source pixel\nvalue.\n" }, { "name": "high", "content": "Set high limit for changing source pixel. Default is 65535. Allowed range is from 0 to\n65535. This option controls maximum possible value that will increase source pixel\nvalue.\n" }, { "name": "planes", "content": "Set which planes to filter. Default is all. Allowed range is from 0 to 15.\n\nCommands\n\nThis filter supports the following commands that corresponds to option of same name:\n" }, { "name": "factor", "content": "" }, { "name": "threshold", "content": "" }, { "name": "tolerance", "content": "" }, { "name": "low", "content": "" }, { "name": "high", "content": "" }, { "name": "planes", "content": "" }, { "name": "ass", "content": "Same as the subtitles filter, except that it doesn't require libavcodec and libavformat to\nwork. On the other hand, it is limited to ASS (Advanced Substation Alpha) subtitles files.\n\nThis filter accepts the following option in addition to the common options from the subtitles\nfilter:\n" }, { "name": "shaping", "content": "Set the shaping engine\n\nAvailable values are:\n\nauto\nThe default libass shaping engine, which is the best available.\n\nsimple\nFast, font-agnostic shaper that can do only substitutions\n\ncomplex\nSlower shaper using OpenType for substitutions and positioning\n\nThe default is \"auto\".\n" }, { "name": "atadenoise", "content": "Apply an Adaptive Temporal Averaging Denoiser to the video input.\n\nThe filter accepts the following options:\n\n0a Set threshold A for 1st plane. Default is 0.02. Valid range is 0 to 0.3.\n\n0b Set threshold B for 1st plane. Default is 0.04. Valid range is 0 to 5.\n\n1a Set threshold A for 2nd plane. Default is 0.02. Valid range is 0 to 0.3.\n\n1b Set threshold B for 2nd plane. Default is 0.04. Valid range is 0 to 5.\n\n2a Set threshold A for 3rd plane. Default is 0.02. Valid range is 0 to 0.3.\n\n2b Set threshold B for 3rd plane. Default is 0.04. Valid range is 0 to 5.\n\nThreshold A is designed to react on abrupt changes in the input signal and threshold B is\ndesigned to react on continuous changes in the input signal.\n\ns Set number of frames filter will use for averaging. Default is 9. Must be odd number in\nrange [5, 129].\n\np Set what planes of frame filter will use for averaging. Default is all.\n\na Set what variant of algorithm filter will use for averaging. Default is \"p\" parallel.\nAlternatively can be set to \"s\" serial.\n\nParallel can be faster then serial, while other way around is never true. Parallel will\nabort early on first change being greater then thresholds, while serial will continue\nprocessing other side of frames if they are equal or below thresholds.\n\n0s\n1s\n2s Set sigma for 1st plane, 2nd plane or 3rd plane. Default is 32767. Valid range is from 0\nto 32767. This options controls weight for each pixel in radius defined by size.\nDefault value means every pixel have same weight. Setting this option to 0 effectively\ndisables filtering.\n\nCommands\n\nThis filter supports same commands as options except option \"s\". The command accepts the\nsame syntax of the corresponding option.\n" }, { "name": "avgblur", "content": "Apply average blur filter.\n\nThe filter accepts the following options:\n" }, { "name": "sizeX", "content": "Set horizontal radius size.\n" }, { "name": "planes", "content": "Set which planes to filter. By default all planes are filtered.\n" }, { "name": "sizeY", "content": "Set vertical radius size, if zero it will be same as \"sizeX\". Default is 0.\n\nCommands\n\nThis filter supports same commands as options. The command accepts the same syntax of the\ncorresponding option.\n\nIf the specified expression is not valid, it is kept at its current value.\n" }, { "name": "bbox", "content": "Compute the bounding box for the non-black pixels in the input frame luminance plane.\n\nThis filter computes the bounding box containing all the pixels with a luminance value\ngreater than the minimum allowed value. The parameters describing the bounding box are\nprinted on the filter log.\n\nThe filter accepts the following option:\n\nminval\nSet the minimal luminance value. Default is 16.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "bilateral", "content": "Apply bilateral filter, spatial smoothing while preserving edges.\n\nThe filter accepts the following options:\n" }, { "name": "sigmaS", "content": "Set sigma of gaussian function to calculate spatial weight. Allowed range is 0 to 512.\nDefault is 0.1.\n" }, { "name": "sigmaR", "content": "Set sigma of gaussian function to calculate range weight. Allowed range is 0 to 1.\nDefault is 0.1.\n" }, { "name": "planes", "content": "Set planes to filter. Default is first only.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "bitplanenoise", "content": "Show and measure bit plane noise.\n\nThe filter accepts the following options:\n" }, { "name": "bitplane", "content": "Set which plane to analyze. Default is 1.\n" }, { "name": "filter", "content": "Filter out noisy pixels from \"bitplane\" set above. Default is disabled.\n" }, { "name": "blackdetect", "content": "Detect video intervals that are (almost) completely black. Can be useful to detect chapter\ntransitions, commercials, or invalid recordings.\n\nThe filter outputs its detection analysis to both the log as well as frame metadata. If a\nblack segment of at least the specified minimum duration is found, a line with the start and\nend timestamps as well as duration is printed to the log with level \"info\". In addition, a\nlog line with level \"debug\" is printed per frame showing the black amount detected for that\nframe.\n\nThe filter also attaches metadata to the first frame of a black segment with key\n\"lavfi.blackstart\" and to the first frame after the black segment ends with key\n\"lavfi.blackend\". The value is the frame's timestamp. This metadata is added regardless of\nthe minimum duration specified.\n\nThe filter accepts the following options:\n\nblackminduration, d\nSet the minimum detected black duration expressed in seconds. It must be a non-negative\nfloating point number.\n\nDefault value is 2.0.\n\npictureblackratioth, picth\nSet the threshold for considering a picture \"black\". Express the minimum value for the\nratio:\n\n / \n\nfor which a picture is considered black. Default value is 0.98.\n\npixelblackth, pixth\nSet the threshold for considering a pixel \"black\".\n\nThe threshold expresses the maximum pixel luminance value for which a pixel is considered\n\"black\". The provided value is scaled according to the following equation:\n\n = + * \n\nluminancerangesize and luminanceminimumvalue depend on the input video format, the\nrange is [0-255] for YUV full-range formats and [16-235] for YUV non full-range formats.\n\nDefault value is 0.10.\n\nThe following example sets the maximum pixel threshold to the minimum value, and detects only\nblack intervals of 2 or more seconds:\n\nblackdetect=d=2:pixth=0.00\n" }, { "name": "blackframe", "content": "Detect frames that are (almost) completely black. Can be useful to detect chapter transitions\nor commercials. Output lines consist of the frame number of the detected frame, the\npercentage of blackness, the position in the file if known or -1 and the timestamp in\nseconds.\n\nIn order to display the output lines, you need to set the loglevel at least to the\nAVLOGINFO value.\n\nThis filter exports frame metadata \"lavfi.blackframe.pblack\". The value represents the\npercentage of pixels in the picture that are below the threshold value.\n\nIt accepts the following parameters:\n" }, { "name": "amount", "content": "The percentage of the pixels that have to be below the threshold; it defaults to 98.\n" }, { "name": "threshold, thresh", "content": "The threshold below which a pixel value is considered black; it defaults to 32.\n" }, { "name": "blend", "content": "Blend two video frames into each other.\n\nThe \"blend\" filter takes two input streams and outputs one stream, the first input is the\n\"top\" layer and second input is \"bottom\" layer. By default, the output terminates when the\nlongest input terminates.\n\nThe \"tblend\" (time blend) filter takes two consecutive frames from one single stream, and\noutputs the result obtained by blending the new frame on top of the old frame.\n\nA description of the accepted options follows.\n\nc0mode\nc1mode\nc2mode\nc3mode\nallmode\nSet blend mode for specific pixel component or all pixel components in case of allmode.\nDefault value is \"normal\".\n\nAvailable values for component modes are:\n\naddition\ngrainmerge\nand\naverage\nburn\ndarken\ndifference\ngrainextract\ndivide\ndodge\nfreeze\nexclusion\nextremity\nglow\nhardlight\nhardmix\nheat\nlighten\nlinearlight\nmultiply\nmultiply128\nnegation\nnormal\nor\noverlay\nphoenix\npinlight\nreflect\nscreen\nsoftlight\nsubtract\nvividlight\nxor\nc0opacity\nc1opacity\nc2opacity\nc3opacity\nallopacity\nSet blend opacity for specific pixel component or all pixel components in case of\nallopacity. Only used in combination with pixel component blend modes.\n\nc0expr\nc1expr\nc2expr\nc3expr\nallexpr\nSet blend expression for specific pixel component or all pixel components in case of\nallexpr. Note that related mode options will be ignored if those are set.\n\nThe expressions can use the following variables:\n\nN The sequential number of the filtered frame, starting from 0.\n\nX\nY the coordinates of the current sample\n\nW\nH the width and height of currently filtered plane\n\nSW\nSH Width and height scale for the plane being filtered. It is the ratio between the\ndimensions of the current plane to the luma plane, e.g. for a \"yuv420p\" frame, the\nvalues are \"1,1\" for the luma plane and \"0.5,0.5\" for the chroma planes.\n\nT Time of the current frame, expressed in seconds.\n\nTOP, A\nValue of pixel component at current location for first video frame (top layer).\n\nBOTTOM, B\nValue of pixel component at current location for second video frame (bottom layer).\n\nThe \"blend\" filter also supports the framesync options.\n\nExamples\n\n• Apply transition from bottom layer to top layer in first 10 seconds:\n\nblend=allexpr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'\n\n• Apply linear horizontal transition from top layer to bottom layer:\n\nblend=allexpr='A*(X/W)+B*(1-X/W)'\n\n• Apply 1x1 checkerboard effect:\n\nblend=allexpr='if(eq(mod(X,2),mod(Y,2)),A,B)'\n\n• Apply uncover left effect:\n\nblend=allexpr='if(gte(N*SW+X,W),A,B)'\n\n• Apply uncover down effect:\n\nblend=allexpr='if(gte(Y-N*SH,0),A,B)'\n\n• Apply uncover up-left effect:\n\nblend=allexpr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'\n\n• Split diagonally video and shows top and bottom layer on each side:\n\nblend=allexpr='if(gt(X,Y*(W/H)),A,B)'\n\n• Display differences between the current and the previous frame:\n\ntblend=allmode=grainextract\n\nCommands\n\nThis filter supports same commands as options.\n" }, { "name": "bm3d", "content": "Denoise frames using Block-Matching 3D algorithm.\n\nThe filter accepts the following options.\n" }, { "name": "sigma", "content": "Set denoising strength. Default value is 1. Allowed range is from 0 to 999.9. The\ndenoising algorithm is very sensitive to sigma, so adjust it according to the source.\n" }, { "name": "block", "content": "Set local patch size. This sets dimensions in 2D.\n" }, { "name": "bstep", "content": "Set sliding step for processing blocks. Default value is 4. Allowed range is from 1 to\n64. Smaller values allows processing more reference blocks and is slower.\n" }, { "name": "group", "content": "Set maximal number of similar blocks for 3rd dimension. Default value is 1. When set to\n1, no block matching is done. Larger values allows more blocks in single group. Allowed\nrange is from 1 to 256.\n" }, { "name": "range", "content": "Set radius for search block matching. Default is 9. Allowed range is from 1 to\nINT32MAX.\n" }, { "name": "mstep", "content": "Set step between two search locations for block matching. Default is 1. Allowed range is\nfrom 1 to 64. Smaller is slower.\n" }, { "name": "thmse", "content": "Set threshold of mean square error for block matching. Valid range is 0 to INT32MAX.\n" }, { "name": "hdthr", "content": "Set thresholding parameter for hard thresholding in 3D transformed domain. Larger values\nresults in stronger hard-thresholding filtering in frequency domain.\n" }, { "name": "estim", "content": "Set filtering estimation mode. Can be \"basic\" or \"final\". Default is \"basic\".\n\nref If enabled, filter will use 2nd stream for block matching. Default is disabled for\n\"basic\" value of estim option, and always enabled if value of estim is \"final\".\n" }, { "name": "planes", "content": "Set planes to filter. Default is all available except alpha.\n\nExamples\n\n• Basic filtering with bm3d:\n\nbm3d=sigma=3:block=4:bstep=2:group=1:estim=basic\n\n• Same as above, but filtering only luma:\n\nbm3d=sigma=3:block=4:bstep=2:group=1:estim=basic:planes=1\n\n• Same as above, but with both estimation modes:\n\nsplit[a][b],[a]bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1\n\n• Same as above, but prefilter with nlmeans filter instead:\n\nsplit[a][b],[a]nlmeans=s=3:r=7:p=3[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1\n" }, { "name": "boxblur", "content": "Apply a boxblur algorithm to the input video.\n\nIt accepts the following parameters:\n\nlumaradius, lr\nlumapower, lp\nchromaradius, cr\nchromapower, cp\nalpharadius, ar\nalphapower, ap\n\nA description of the accepted options follows.\n\nlumaradius, lr\nchromaradius, cr\nalpharadius, ar\nSet an expression for the box radius in pixels used for blurring the corresponding input\nplane.\n\nThe radius value must be a non-negative number, and must not be greater than the value of\nthe expression \"min(w,h)/2\" for the luma and alpha planes, and of \"min(cw,ch)/2\" for the\nchroma planes.\n\nDefault value for lumaradius is \"2\". If not specified, chromaradius and alpharadius\ndefault to the corresponding value set for lumaradius.\n\nThe expressions can contain the following constants:\n\nw\nh The input width and height in pixels.\n\ncw\nch The input chroma image width and height in pixels.\n\nhsub\nvsub\nThe horizontal and vertical chroma subsample values. For example, for the pixel\nformat \"yuv422p\", hsub is 2 and vsub is 1.\n\nlumapower, lp\nchromapower, cp\nalphapower, ap\nSpecify how many times the boxblur filter is applied to the corresponding plane.\n\nDefault value for lumapower is 2. If not specified, chromapower and alphapower default\nto the corresponding value set for lumapower.\n\nA value of 0 will disable the effect.\n\nExamples\n\n• Apply a boxblur filter with the luma, chroma, and alpha radii set to 2:\n\nboxblur=lumaradius=2:lumapower=1\nboxblur=2:1\n\n• Set the luma radius to 2, and alpha and chroma radius to 0:\n\nboxblur=2:1:cr=0:ar=0\n\n• Set the luma and chroma radii to a fraction of the video dimension:\n\nboxblur=lumaradius=min(h\\,w)/10:lumapower=1:chromaradius=min(cw\\,ch)/10:chromapower=1\n" }, { "name": "bwdif", "content": "Deinterlace the input video (\"bwdif\" stands for \"Bob Weaver Deinterlacing Filter\").\n\nMotion adaptive deinterlacing based on yadif with the use of w3fdif and cubic interpolation\nalgorithms. It accepts the following parameters:\n" }, { "name": "mode", "content": "The interlacing mode to adopt. It accepts one of the following values:\n\n0, sendframe\nOutput one frame for each frame.\n\n1, sendfield\nOutput one frame for each field.\n\nThe default value is \"sendfield\".\n" }, { "name": "parity", "content": "The picture field parity assumed for the input interlaced video. It accepts one of the\nfollowing values:\n\n0, tff\nAssume the top field is first.\n\n1, bff\nAssume the bottom field is first.\n\n-1, auto\nEnable automatic detection of field parity.\n\nThe default value is \"auto\". If the interlacing is unknown or the decoder does not\nexport this information, top field first will be assumed.\n" }, { "name": "deint", "content": "Specify which frames to deinterlace. Accepts one of the following values:\n\n0, all\nDeinterlace all frames.\n\n1, interlaced\nOnly deinterlace frames marked as interlaced.\n\nThe default value is \"all\".\n" }, { "name": "cas", "content": "Apply Contrast Adaptive Sharpen filter to video stream.\n\nThe filter accepts the following options:\n" }, { "name": "strength", "content": "Set the sharpening strength. Default value is 0.\n" }, { "name": "planes", "content": "Set planes to filter. Default value is to filter all planes except alpha plane.\n\nCommands\n\nThis filter supports same commands as options.\n" }, { "name": "chromahold", "content": "Remove all color information for all colors except for certain one.\n\nThe filter accepts the following options:\n" }, { "name": "color", "content": "The color which will not be replaced with neutral chroma.\n" }, { "name": "similarity", "content": "Similarity percentage with the above color. 0.01 matches only the exact key color, while\n1.0 matches everything.\n" }, { "name": "blend", "content": "Blend percentage. 0.0 makes pixels either fully gray, or not gray at all. Higher values\nresult in more preserved color.\n\nyuv Signals that the color passed is already in YUV instead of RGB.\n\nLiteral colors like \"green\" or \"red\" don't make sense with this enabled anymore. This\ncan be used to pass exact YUV values as hexadecimal numbers.\n\nCommands\n\nThis filter supports same commands as options. The command accepts the same syntax of the\ncorresponding option.\n\nIf the specified expression is not valid, it is kept at its current value.\n" }, { "name": "chromakey", "content": "YUV colorspace color/chroma keying.\n\nThe filter accepts the following options:\n" }, { "name": "color", "content": "The color which will be replaced with transparency.\n" }, { "name": "similarity", "content": "Similarity percentage with the key color.\n\n0.01 matches only the exact key color, while 1.0 matches everything.\n" }, { "name": "blend", "content": "Blend percentage.\n\n0.0 makes pixels either fully transparent, or not transparent at all.\n\nHigher values result in semi-transparent pixels, with a higher transparency the more\nsimilar the pixels color is to the key color.\n\nyuv Signals that the color passed is already in YUV instead of RGB.\n\nLiteral colors like \"green\" or \"red\" don't make sense with this enabled anymore. This\ncan be used to pass exact YUV values as hexadecimal numbers.\n\nCommands\n\nThis filter supports same commands as options. The command accepts the same syntax of the\ncorresponding option.\n\nIf the specified expression is not valid, it is kept at its current value.\n\nExamples\n\n• Make every green pixel in the input image transparent:\n\nffmpeg -i input.png -vf chromakey=green out.png\n\n• Overlay a greenscreen-video on top of a static black background.\n\nffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filtercomplex \"[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]\" -map \"[out]\" output.mkv\n" }, { "name": "chromanr", "content": "Reduce chrominance noise.\n\nThe filter accepts the following options:\n" }, { "name": "thres", "content": "Set threshold for averaging chrominance values. Sum of absolute difference of Y, U and V\npixel components of current pixel and neighbour pixels lower than this threshold will be\nused in averaging. Luma component is left unchanged and is copied to output. Default\nvalue is 30. Allowed range is from 1 to 200.\n" }, { "name": "sizew", "content": "Set horizontal radius of rectangle used for averaging. Allowed range is from 1 to 100.\nDefault value is 5.\n" }, { "name": "sizeh", "content": "Set vertical radius of rectangle used for averaging. Allowed range is from 1 to 100.\nDefault value is 5.\n" }, { "name": "stepw", "content": "Set horizontal step when averaging. Default value is 1. Allowed range is from 1 to 50.\nMostly useful to speed-up filtering.\n" }, { "name": "steph", "content": "Set vertical step when averaging. Default value is 1. Allowed range is from 1 to 50.\nMostly useful to speed-up filtering.\n" }, { "name": "threy", "content": "Set Y threshold for averaging chrominance values. Set finer control for max allowed\ndifference between Y components of current pixel and neigbour pixels. Default value is\n200. Allowed range is from 1 to 200.\n" }, { "name": "threu", "content": "Set U threshold for averaging chrominance values. Set finer control for max allowed\ndifference between U components of current pixel and neigbour pixels. Default value is\n200. Allowed range is from 1 to 200.\n" }, { "name": "threv", "content": "Set V threshold for averaging chrominance values. Set finer control for max allowed\ndifference between V components of current pixel and neigbour pixels. Default value is\n200. Allowed range is from 1 to 200.\n\nCommands\n\nThis filter supports same commands as options. The command accepts the same syntax of the\ncorresponding option.\n" }, { "name": "chromashift", "content": "Shift chroma pixels horizontally and/or vertically.\n\nThe filter accepts the following options:\n\ncbh Set amount to shift chroma-blue horizontally.\n\ncbv Set amount to shift chroma-blue vertically.\n\ncrh Set amount to shift chroma-red horizontally.\n\ncrv Set amount to shift chroma-red vertically.\n" }, { "name": "edge", "content": "Set edge mode, can be smear, default, or warp.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "ciescope", "content": "Display CIE color diagram with pixels overlaid onto it.\n\nThe filter accepts the following options:\n" }, { "name": "system", "content": "Set color system.\n\nntsc, 470m\nebu, 470bg\nsmpte\n240m\napple\nwidergb\ncie1931\nrec709, hdtv\nuhdtv, rec2020\ndcip3\ncie Set CIE system.\n\nxyy\nucs\nluv" }, { "name": "gamuts", "content": "Set what gamuts to draw.\n\nSee \"system\" option for available values.\n" }, { "name": "size, s", "content": "Set ciescope size, by default set to 512.\n" }, { "name": "intensity, i", "content": "Set intensity used to map input pixel values to CIE diagram.\n" }, { "name": "contrast", "content": "Set contrast used to draw tongue colors that are out of active color system gamut.\n" }, { "name": "corrgamma", "content": "Correct gamma displayed on scope, by default enabled.\n" }, { "name": "showwhite", "content": "Show white point on CIE diagram, by default disabled.\n" }, { "name": "gamma", "content": "Set input gamma. Used only with XYZ input color space.\n" }, { "name": "codecview", "content": "Visualize information exported by some codecs.\n\nSome codecs can export information through frames using side-data or other means. For\nexample, some MPEG based codecs export motion vectors through the exportmvs flag in the\ncodec flags2 option.\n\nThe filter accepts the following option:\n\nmv Set motion vectors to visualize.\n\nAvailable flags for mv are:\n\npf forward predicted MVs of P-frames\n\nbf forward predicted MVs of B-frames\n\nbb backward predicted MVs of B-frames\n\nqp Display quantization parameters using the chroma planes.\n\nmvtype, mvt\nSet motion vectors type to visualize. Includes MVs from all frames unless specified by\nframetype option.\n\nAvailable flags for mvtype are:\n\nfp forward predicted MVs\n\nbp backward predicted MVs\n\nframetype, ft\nSet frame type to visualize motion vectors of.\n\nAvailable flags for frametype are:\n\nif intra-coded frames (I-frames)\n\npf predicted frames (P-frames)\n\nbf bi-directionally predicted frames (B-frames)\n\nExamples\n\n• Visualize forward predicted MVs of all frames using ffplay:\n\nffplay -flags2 +exportmvs input.mp4 -vf codecview=mvtype=fp\n\n• Visualize multi-directionals MVs of P and B-Frames using ffplay:\n\nffplay -flags2 +exportmvs input.mp4 -vf codecview=mv=pf+bf+bb\n" }, { "name": "colorbalance", "content": "Modify intensity of primary colors (red, green and blue) of input frames.\n\nThe filter allows an input frame to be adjusted in the shadows, midtones or highlights\nregions for the red-cyan, green-magenta or blue-yellow balance.\n\nA positive adjustment value shifts the balance towards the primary color, a negative value\ntowards the complementary color.\n\nThe filter accepts the following options:\n\nrs\ngs\nbs Adjust red, green and blue shadows (darkest pixels).\n\nrm\ngm\nbm Adjust red, green and blue midtones (medium pixels).\n\nrh\ngh\nbh Adjust red, green and blue highlights (brightest pixels).\n\nAllowed ranges for options are \"[-1.0, 1.0]\". Defaults are 0.\n\npl Preserve lightness when changing color balance. Default is disabled.\n\nExamples\n\n• Add red color cast to shadows:\n\ncolorbalance=rs=.3\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "colorcontrast", "content": "Adjust color contrast between RGB components.\n\nThe filter accepts the following options:\n\nrc Set the red-cyan contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0.\n\ngm Set the green-magenta contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0.\n\nby Set the blue-yellow contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0.\n" }, { "name": "rcw", "content": "" }, { "name": "gmw", "content": "byw Set the weight of each \"rc\", \"gm\", \"by\" option value. Default value is 0.0. Allowed\nrange is from 0.0 to 1.0. If all weights are 0.0 filtering is disabled.\n\npl Set the amount of preserving lightness. Default value is 0.0. Allowed range is from 0.0\nto 1.0.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "colorcorrect", "content": "Adjust color white balance selectively for blacks and whites. This filter operates in YUV\ncolorspace.\n\nThe filter accepts the following options:\n\nrl Set the red shadow spot. Allowed range is from -1.0 to 1.0. Default value is 0.\n\nbl Set the blue shadow spot. Allowed range is from -1.0 to 1.0. Default value is 0.\n\nrh Set the red highlight spot. Allowed range is from -1.0 to 1.0. Default value is 0.\n\nbh Set the red highlight spot. Allowed range is from -1.0 to 1.0. Default value is 0.\n" }, { "name": "saturation", "content": "Set the amount of saturation. Allowed range is from -3.0 to 3.0. Default value is 1.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "colorchannelmixer", "content": "Adjust video input frames by re-mixing color channels.\n\nThis filter modifies a color channel by adding the values associated to the other channels of\nthe same pixels. For example if the value to modify is red, the output value will be:\n\n=* + * + * + *\n\nThe filter accepts the following options:\n\nrr\nrg\nrb\nra Adjust contribution of input red, green, blue and alpha channels for output red channel.\nDefault is 1 for rr, and 0 for rg, rb and ra.\n\ngr\ngg\ngb\nga Adjust contribution of input red, green, blue and alpha channels for output green\nchannel. Default is 1 for gg, and 0 for gr, gb and ga.\n\nbr\nbg\nbb\nba Adjust contribution of input red, green, blue and alpha channels for output blue channel.\nDefault is 1 for bb, and 0 for br, bg and ba.\n\nar\nag\nab\naa Adjust contribution of input red, green, blue and alpha channels for output alpha\nchannel. Default is 1 for aa, and 0 for ar, ag and ab.\n\nAllowed ranges for options are \"[-2.0, 2.0]\".\n\npl Preserve lightness when changing colors. Allowed range is from \"[0.0, 1.0]\". Default is\n0.0, thus disabled.\n\nExamples\n\n• Convert source to grayscale:\n\ncolorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3\n\n• Simulate sepia tones:\n\ncolorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "colorize", "content": "Overlay a solid color on the video stream.\n\nThe filter accepts the following options:\n\nhue Set the color hue. Allowed range is from 0 to 360. Default value is 0.\n" }, { "name": "saturation", "content": "Set the color saturation. Allowed range is from 0 to 1. Default value is 0.5.\n" }, { "name": "lightness", "content": "Set the color lightness. Allowed range is from 0 to 1. Default value is 0.5.\n\nmix Set the mix of source lightness. By default is set to 1.0. Allowed range is from 0.0 to\n1.0.\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "colorkey", "content": "RGB colorspace color keying.\n\nThe filter accepts the following options:\n" }, { "name": "color", "content": "The color which will be replaced with transparency.\n" }, { "name": "similarity", "content": "Similarity percentage with the key color.\n\n0.01 matches only the exact key color, while 1.0 matches everything.\n" }, { "name": "blend", "content": "Blend percentage.\n\n0.0 makes pixels either fully transparent, or not transparent at all.\n\nHigher values result in semi-transparent pixels, with a higher transparency the more\nsimilar the pixels color is to the key color.\n\nExamples\n\n• Make every green pixel in the input image transparent:\n\nffmpeg -i input.png -vf colorkey=green out.png\n\n• Overlay a greenscreen-video on top of a static background image.\n\nffmpeg -i background.png -i video.mp4 -filtercomplex \"[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]\" -map \"[out]\" output.flv\n\nCommands\n\nThis filter supports same commands as options. The command accepts the same syntax of the\ncorresponding option.\n\nIf the specified expression is not valid, it is kept at its current value.\n" }, { "name": "colorhold", "content": "Remove all color information for all RGB colors except for certain one.\n\nThe filter accepts the following options:\n" }, { "name": "color", "content": "The color which will not be replaced with neutral gray.\n" }, { "name": "similarity", "content": "Similarity percentage with the above color. 0.01 matches only the exact key color, while\n1.0 matches everything.\n" }, { "name": "blend", "content": "Blend percentage. 0.0 makes pixels fully gray. Higher values result in more preserved\ncolor.\n\nCommands\n\nThis filter supports same commands as options. The command accepts the same syntax of the\ncorresponding option.\n\nIf the specified expression is not valid, it is kept at its current value.\n" }, { "name": "colorlevels", "content": "Adjust video input frames using levels.\n\nThe filter accepts the following options:\n" }, { "name": "rimin", "content": "" }, { "name": "gimin", "content": "" }, { "name": "bimin", "content": "" }, { "name": "aimin", "content": "Adjust red, green, blue and alpha input black point. Allowed ranges for options are\n\"[-1.0, 1.0]\". Defaults are 0.\n" }, { "name": "rimax", "content": "" }, { "name": "gimax", "content": "" }, { "name": "bimax", "content": "" }, { "name": "aimax", "content": "Adjust red, green, blue and alpha input white point. Allowed ranges for options are\n\"[-1.0, 1.0]\". Defaults are 1.\n\nInput levels are used to lighten highlights (bright tones), darken shadows (dark tones),\nchange the balance of bright and dark tones.\n" }, { "name": "romin", "content": "" }, { "name": "gomin", "content": "" }, { "name": "bomin", "content": "" }, { "name": "aomin", "content": "Adjust red, green, blue and alpha output black point. Allowed ranges for options are\n\"[0, 1.0]\". Defaults are 0.\n" }, { "name": "romax", "content": "" }, { "name": "gomax", "content": "" }, { "name": "bomax", "content": "" }, { "name": "aomax", "content": "Adjust red, green, blue and alpha output white point. Allowed ranges for options are\n\"[0, 1.0]\". Defaults are 1.\n\nOutput levels allows manual selection of a constrained output level range.\n\nExamples\n\n• Make video output darker:\n\ncolorlevels=rimin=0.058:gimin=0.058:bimin=0.058\n\n• Increase contrast:\n\ncolorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96\n\n• Make video output lighter:\n\ncolorlevels=rimax=0.902:gimax=0.902:bimax=0.902\n\n• Increase brightness:\n\ncolorlevels=romin=0.5:gomin=0.5:bomin=0.5\n\nCommands\n\nThis filter supports the all above options as commands.\n" }, { "name": "colormatrix", "content": "Convert color matrix.\n\nThe filter accepts the following options:\n" }, { "name": "src", "content": "dst Specify the source and destination color matrix. Both values must be specified.\n\nThe accepted values are:\n\nbt709\nBT.709\n\nfcc FCC\n\nbt601\nBT.601\n\nbt470\nBT.470\n\nbt470bg\nBT.470BG\n\nsmpte170m\nSMPTE-170M\n\nsmpte240m\nSMPTE-240M\n\nbt2020\nBT.2020\n\nFor example to convert from BT.601 to SMPTE-240M, use the command:\n\ncolormatrix=bt601:smpte240m\n" }, { "name": "colorspace", "content": "Convert colorspace, transfer characteristics or color primaries. Input video needs to have\nan even size.\n\nThe filter accepts the following options:\n\nall Specify all color properties at once.\n\nThe accepted values are:\n\nbt470m\nBT.470M\n\nbt470bg\nBT.470BG\n\nbt601-6-525\nBT.601-6 525\n\nbt601-6-625\nBT.601-6 625\n\nbt709\nBT.709\n\nsmpte170m\nSMPTE-170M\n\nsmpte240m\nSMPTE-240M\n\nbt2020\nBT.2020\n" }, { "name": "space", "content": "Specify output colorspace.\n\nThe accepted values are:\n\nbt709\nBT.709\n\nfcc FCC\n\nbt470bg\nBT.470BG or BT.601-6 625\n\nsmpte170m\nSMPTE-170M or BT.601-6 525\n\nsmpte240m\nSMPTE-240M\n\nycgco\nYCgCo\n\nbt2020ncl\nBT.2020 with non-constant luminance\n\ntrc Specify output transfer characteristics.\n\nThe accepted values are:\n\nbt709\nBT.709\n\nbt470m\nBT.470M\n\nbt470bg\nBT.470BG\n\ngamma22\nConstant gamma of 2.2\n\ngamma28\nConstant gamma of 2.8\n\nsmpte170m\nSMPTE-170M, BT.601-6 625 or BT.601-6 525\n\nsmpte240m\nSMPTE-240M\n\nsrgb\nSRGB\n\niec61966-2-1\niec61966-2-1\n\niec61966-2-4\niec61966-2-4\n\nxvycc\nxvycc\n\nbt2020-10\nBT.2020 for 10-bits content\n\nbt2020-12\nBT.2020 for 12-bits content\n" }, { "name": "primaries", "content": "Specify output color primaries.\n\nThe accepted values are:\n\nbt709\nBT.709\n\nbt470m\nBT.470M\n\nbt470bg\nBT.470BG or BT.601-6 625\n\nsmpte170m\nSMPTE-170M or BT.601-6 525\n\nsmpte240m\nSMPTE-240M\n\nfilm\nfilm\n\nsmpte431\nSMPTE-431\n\nsmpte432\nSMPTE-432\n\nbt2020\nBT.2020\n\njedec-p22\nJEDEC P22 phosphors\n" }, { "name": "range", "content": "Specify output color range.\n\nThe accepted values are:\n\ntv TV (restricted) range\n\nmpeg\nMPEG (restricted) range\n\npc PC (full) range\n\njpeg\nJPEG (full) range\n" }, { "name": "format", "content": "Specify output color format.\n\nThe accepted values are:\n\nyuv420p\nYUV 4:2:0 planar 8-bits\n\nyuv420p10\nYUV 4:2:0 planar 10-bits\n\nyuv420p12\nYUV 4:2:0 planar 12-bits\n\nyuv422p\nYUV 4:2:2 planar 8-bits\n\nyuv422p10\nYUV 4:2:2 planar 10-bits\n\nyuv422p12\nYUV 4:2:2 planar 12-bits\n\nyuv444p\nYUV 4:4:4 planar 8-bits\n\nyuv444p10\nYUV 4:4:4 planar 10-bits\n\nyuv444p12\nYUV 4:4:4 planar 12-bits\n" }, { "name": "fast", "content": "Do a fast conversion, which skips gamma/primary correction. This will take significantly\nless CPU, but will be mathematically incorrect. To get output compatible with that\nproduced by the colormatrix filter, use fast=1.\n" }, { "name": "dither", "content": "Specify dithering mode.\n\nThe accepted values are:\n\nnone\nNo dithering\n\nfsb Floyd-Steinberg dithering\n" }, { "name": "wpadapt", "content": "Whitepoint adaptation mode.\n\nThe accepted values are:\n\nbradford\nBradford whitepoint adaptation\n\nvonkries\nvon Kries whitepoint adaptation\n\nidentity\nidentity whitepoint adaptation (i.e. no whitepoint adaptation)\n" }, { "name": "iall", "content": "Override all input properties at once. Same accepted values as all.\n" }, { "name": "ispace", "content": "Override input colorspace. Same accepted values as space.\n" }, { "name": "iprimaries", "content": "Override input color primaries. Same accepted values as primaries.\n" }, { "name": "itrc", "content": "Override input transfer characteristics. Same accepted values as trc.\n" }, { "name": "irange", "content": "Override input color range. Same accepted values as range.\n\nThe filter converts the transfer characteristics, color space and color primaries to the\nspecified user values. The output value, if not specified, is set to a default value based on\nthe \"all\" property. If that property is also not specified, the filter will log an error. The\noutput color range and format default to the same value as the input color range and format.\nThe input transfer characteristics, color space, color primaries and color range should be\nset on the input data. If any of these are missing, the filter will log an error and no\nconversion will take place.\n\nFor example to convert the input to SMPTE-240M, use the command:\n\ncolorspace=smpte240m\n" }, { "name": "colortemperature", "content": "Adjust color temperature in video to simulate variations in ambient color temperature.\n\nThe filter accepts the following options:\n" }, { "name": "temperature", "content": "Set the temperature in Kelvin. Allowed range is from 1000 to 40000. Default value is\n6500 K.\n\nmix Set mixing with filtered output. Allowed range is from 0 to 1. Default value is 1.\n\npl Set the amount of preserving lightness. Allowed range is from 0 to 1. Default value is\n0.\n\nCommands\n\nThis filter supports same commands as options.\n" }, { "name": "convolution", "content": "Apply convolution of 3x3, 5x5, 7x7 or horizontal/vertical up to 49 elements.\n\nThe filter accepts the following options:\n\n0m\n1m\n2m\n3m Set matrix for each plane. Matrix is sequence of 9, 25 or 49 signed integers in square\nmode, and from 1 to 49 odd number of signed integers in row mode.\n" }, { "name": "0rdiv", "content": "" }, { "name": "1rdiv", "content": "" }, { "name": "2rdiv", "content": "" }, { "name": "3rdiv", "content": "Set multiplier for calculated value for each plane. If unset or 0, it will be sum of all\nmatrix elements.\n" }, { "name": "0bias", "content": "" }, { "name": "1bias", "content": "" }, { "name": "2bias", "content": "" }, { "name": "3bias", "content": "Set bias for each plane. This value is added to the result of the multiplication. Useful\nfor making the overall image brighter or darker. Default is 0.0.\n" }, { "name": "0mode", "content": "" }, { "name": "1mode", "content": "" }, { "name": "2mode", "content": "" }, { "name": "3mode", "content": "Set matrix mode for each plane. Can be square, row or column. Default is square.\n\nCommands\n\nThis filter supports the all above options as commands.\n\nExamples\n\n• Apply sharpen:\n\nconvolution=\"0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0\"\n\n• Apply blur:\n\nconvolution=\"1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9\"\n\n• Apply edge enhance:\n\nconvolution=\"0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128\"\n\n• Apply edge detect:\n\nconvolution=\"0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128\"\n\n• Apply laplacian edge detector which includes diagonals:\n\nconvolution=\"1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0\"\n\n• Apply emboss:\n\nconvolution=\"-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2\"\n" }, { "name": "convolve", "content": "Apply 2D convolution of video stream in frequency domain using second stream as impulse.\n\nThe filter accepts the following options:\n" }, { "name": "planes", "content": "Set which planes to process.\n" }, { "name": "impulse", "content": "Set which impulse video frames will be processed, can be first or all. Default is all.\n\nThe \"convolve\" filter also supports the framesync options.\n" }, { "name": "copy", "content": "Copy the input video source unchanged to the output. This is mainly useful for testing\npurposes.\n" }, { "name": "coreimage", "content": "Video filtering on GPU using Apple's CoreImage API on OSX.\n\nHardware acceleration is based on an OpenGL context. Usually, this means it is processed by\nvideo hardware. However, software-based OpenGL implementations exist which means there is no\nguarantee for hardware processing. It depends on the respective OSX.\n\nThere are many filters and image generators provided by Apple that come with a large variety\nof options. The filter has to be referenced by its name along with its options.\n\nThe coreimage filter accepts the following options:\n\nlistfilters\nList all available filters and generators along with all their respective options as well\nas possible minimum and maximum values along with the default values.\n\nlistfilters=true\n" }, { "name": "filter", "content": "Specify all filters by their respective name and options. Use listfilters to determine\nall valid filter names and options. Numerical options are specified by a float value and\nare automatically clamped to their respective value range. Vector and color options have\nto be specified by a list of space separated float values. Character escaping has to be\ndone. A special option name \"default\" is available to use default options for a filter.\n\nIt is required to specify either \"default\" or at least one of the filter options. All\nomitted options are used with their default values. The syntax of the filter string is\nas follows:\n\nfilter=@