DC_offset
Min_level
Max_level
+Min_difference
+Max_difference
+Mean_difference
Peak_level
RMS_peak
RMS_trough
Crest_factor
Flat_factor
Peak_count
+Bit_depth
and for Overall:
DC_offset
Min_level
Max_level
+Min_difference
+Max_difference
+Mean_difference
Peak_level
RMS_level
RMS_peak
RMS_trough
Flat_factor
Peak_count
+Bit_depth
Number_of_samples
For example full key look like this @code{lavfi.astats.1.DC_offset} or
For description what each key means read bellow.
+@item reset
+Set number of frame after which stats are going to be recalculated.
+Default is disabled.
@end table
A description of each shown parameter follows:
@item Max level
Maximal sample level.
+@item Min difference
+Minimal difference between two consecutive samples.
+
+@item Max difference
+Maximal difference between two consecutive samples.
+
+@item Mean difference
+Mean difference between two consecutive samples.
+The average of each difference between two consecutive samples.
+
@item Peak level dB
@item RMS level dB
Standard peak and RMS level measured in dBFS.
@item Peak count
Number of occasions (not the number of samples) that the signal attained either
@var{Min level} or @var{Max level}.
+
+@item Bit depth
+Overall bit depth of audio. Number of bits used for each sample.
@end table
@section astreamsync
To fix a 5.1 WAV improperly encoded in AAC's native channel order
@example
-ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav
+ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav
@end example
@section channelsplit
shorter than the decay time, because the human ear is more sensitive to sudden
loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and
a typical value for decay is 0.8 seconds.
+If specified number of attacks & decays is lower than number of channels, the last
+set attack/decay will be used for all remaining channels.
@item points
A list of points for the transfer function, specified in dB relative to the
used to prevent clipping.
@end table
+@section dynaudnorm
+Dynamic Audio Normalizer.
+
+This filter applies a certain amount of gain to the input audio in order
+to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
+contrast to more "simple" normalization algorithms, the Dynamic Audio
+Normalizer *dynamically* re-adjusts the gain factor to the input audio.
+This allows for applying extra gain to the "quiet" sections of the audio
+while avoiding distortions or clipping the "loud" sections. In other words:
+The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
+sections, in the sense that the volume of each section is brought to the
+same target level. Note, however, that the Dynamic Audio Normalizer achieves
+this goal *without* applying "dynamic range compressing". It will retain 100%
+of the dynamic range *within* each section of the audio file.
+
+@table @option
+@item f
+Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
+Default is 500 milliseconds.
+The Dynamic Audio Normalizer processes the input audio in small chunks,
+referred to as frames. This is required, because a peak magnitude has no
+meaning for just a single sample value. Instead, we need to determine the
+peak magnitude for a contiguous sequence of sample values. While a "standard"
+normalizer would simply use the peak magnitude of the complete file, the
+Dynamic Audio Normalizer determines the peak magnitude individually for each
+frame. The length of a frame is specified in milliseconds. By default, the
+Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
+been found to give good results with most files.
+Note that the exact frame length, in number of samples, will be determined
+automatically, based on the sampling rate of the individual input audio file.
+
+@item g
+Set the Gaussian filter window size. In range from 3 to 301, must be odd
+number. Default is 31.
+Probably the most important parameter of the Dynamic Audio Normalizer is the
+@code{window size} of the Gaussian smoothing filter. The filter's window size
+is specified in frames, centered around the current frame. For the sake of
+simplicity, this must be an odd number. Consequently, the default value of 31
+takes into account the current frame, as well as the 15 preceding frames and
+the 15 subsequent frames. Using a larger window results in a stronger
+smoothing effect and thus in less gain variation, i.e. slower gain
+adaptation. Conversely, using a smaller window results in a weaker smoothing
+effect and thus in more gain variation, i.e. faster gain adaptation.
+In other words, the more you increase this value, the more the Dynamic Audio
+Normalizer will behave like a "traditional" normalization filter. On the
+contrary, the more you decrease this value, the more the Dynamic Audio
+Normalizer will behave like a dynamic range compressor.
+
+@item p
+Set the target peak value. This specifies the highest permissible magnitude
+level for the normalized audio input. This filter will try to approach the
+target peak magnitude as closely as possible, but at the same time it also
+makes sure that the normalized signal will never exceed the peak magnitude.
+A frame's maximum local gain factor is imposed directly by the target peak
+magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
+It is not recommended to go above this value.
+
+@item m
+Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
+The Dynamic Audio Normalizer determines the maximum possible (local) gain
+factor for each input frame, i.e. the maximum gain factor that does not
+result in clipping or distortion. The maximum gain factor is determined by
+the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
+additionally bounds the frame's maximum gain factor by a predetermined
+(global) maximum gain factor. This is done in order to avoid excessive gain
+factors in "silent" or almost silent frames. By default, the maximum gain
+factor is 10.0, For most inputs the default value should be sufficient and
+it usually is not recommended to increase this value. Though, for input
+with an extremely low overall volume level, it may be necessary to allow even
+higher gain factors. Note, however, that the Dynamic Audio Normalizer does
+not simply apply a "hard" threshold (i.e. cut off values above the threshold).
+Instead, a "sigmoid" threshold function will be applied. This way, the
+gain factors will smoothly approach the threshold value, but never exceed that
+value.
+
+@item r
+Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
+By default, the Dynamic Audio Normalizer performs "peak" normalization.
+This means that the maximum local gain factor for each frame is defined
+(only) by the frame's highest magnitude sample. This way, the samples can
+be amplified as much as possible without exceeding the maximum signal
+level, i.e. without clipping. Optionally, however, the Dynamic Audio
+Normalizer can also take into account the frame's root mean square,
+abbreviated RMS. In electrical engineering, the RMS is commonly used to
+determine the power of a time-varying signal. It is therefore considered
+that the RMS is a better approximation of the "perceived loudness" than
+just looking at the signal's peak magnitude. Consequently, by adjusting all
+frames to a constant RMS value, a uniform "perceived loudness" can be
+established. If a target RMS value has been specified, a frame's local gain
+factor is defined as the factor that would result in exactly that RMS value.
+Note, however, that the maximum local gain factor is still restricted by the
+frame's highest magnitude sample, in order to prevent clipping.
+
+@item n
+Enable channels coupling. By default is enabled.
+By default, the Dynamic Audio Normalizer will amplify all channels by the same
+amount. This means the same gain factor will be applied to all channels, i.e.
+the maximum possible gain factor is determined by the "loudest" channel.
+However, in some recordings, it may happen that the volume of the different
+channels is uneven, e.g. one channel may be "quieter" than the other one(s).
+In this case, this option can be used to disable the channel coupling. This way,
+the gain factor will be determined independently for each channel, depending
+only on the individual channel's highest magnitude sample. This allows for
+harmonizing the volume of the different channels.
+
+@item c
+Enable DC bias correction. By default is disabled.
+An audio signal (in the time domain) is a sequence of sample values.
+In the Dynamic Audio Normalizer these sample values are represented in the
+-1.0 to 1.0 range, regardless of the original input format. Normally, the
+audio signal, or "waveform", should be centered around the zero point.
+That means if we calculate the mean value of all samples in a file, or in a
+single frame, then the result should be 0.0 or at least very close to that
+value. If, however, there is a significant deviation of the mean value from
+0.0, in either positive or negative direction, this is referred to as a
+DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
+Audio Normalizer provides optional DC bias correction.
+With DC bias correction enabled, the Dynamic Audio Normalizer will determine
+the mean value, or "DC correction" offset, of each input frame and subtract
+that value from all of the frame's sample values which ensures those samples
+are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
+boundaries, the DC correction offset values will be interpolated smoothly
+between neighbouring frames.
+
+@item b
+Enable alternative boundary mode. By default is disabled.
+The Dynamic Audio Normalizer takes into account a certain neighbourhood
+around each frame. This includes the preceding frames as well as the
+subsequent frames. However, for the "boundary" frames, located at the very
+beginning and at the very end of the audio file, not all neighbouring
+frames are available. In particular, for the first few frames in the audio
+file, the preceding frames are not known. And, similarly, for the last few
+frames in the audio file, the subsequent frames are not known. Thus, the
+question arises which gain factors should be assumed for the missing frames
+in the "boundary" region. The Dynamic Audio Normalizer implements two modes
+to deal with this situation. The default boundary mode assumes a gain factor
+of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
+"fade out" at the beginning and at the end of the input, respectively.
+
+@item s
+Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
+By default, the Dynamic Audio Normalizer does not apply "traditional"
+compression. This means that signal peaks will not be pruned and thus the
+full dynamic range will be retained within each local neighbourhood. However,
+in some cases it may be desirable to combine the Dynamic Audio Normalizer's
+normalization algorithm with a more "traditional" compression.
+For this purpose, the Dynamic Audio Normalizer provides an optional compression
+(thresholding) function. If (and only if) the compression feature is enabled,
+all input frames will be processed by a soft knee thresholding function prior
+to the actual normalization process. Put simply, the thresholding function is
+going to prune all samples whose magnitude exceeds a certain threshold value.
+However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
+value. Instead, the threshold value will be adjusted for each individual
+frame.
+In general, smaller parameters result in stronger compression, and vice versa.
+Values below 3.0 are not recommended, because audible distortion may appear.
+@end table
+
@section earwax
Make audio easier to listen to on headphones.
@code{1}.
@end table
+@section deflate
+
+Apply deflate effect to the video.
+
+This filter replaces the pixel by the local(3x3) average by taking into account
+only values lower than the pixel.
+
+It accepts the following options:
+
+@table @option
+@item threshold0
+@item threshold1
+@item threshold2
+@item threshold3
+Allows to limit the maximum change for each plane, default is 65535.
+If 0, plane will remain unchanged.
+@end table
+
@section dejudder
Remove judder produced by partially interlaced telecined content.
pattern. This is to be used if the stream is cut. The default value is @code{0}.
@end table
+@section dilation
+
+Apply dilation effect to the video.
+
+This filter replaces the pixel by the local(3x3) maximum.
+
+It accepts the following options:
+
+@table @option
+@item threshold0
+@item threshold1
+@item threshold2
+@item threshold3
+Allows to limit the maximum change for each plane, default is 65535.
+If 0, plane will remain unchanged.
+
+@item coordinates
+Flag which specifies the pixel to refer to. Default is 255 i.e. all eight
+pixels are used.
+
+Flags to local 3x3 coordinates maps like this:
+
+ 1 2 3
+ 4 5
+ 6 7 8
+@end table
+
@section drawbox
Draw a colored box on the input image.
@end example
@end itemize
+@section drawgraph, adrawgraph
+
+Draw a graph using input video or audio metadata.
+
+It accepts the following parameters:
+
+@table @option
+@item m1
+Set 1st frame metadata key from which metadata values will be used to draw a graph.
+
+@item fg1
+Set 1st foreground color expression.
+
+@item m2
+Set 2nd frame metadata key from which metadata values will be used to draw a graph.
+
+@item fg2
+Set 2nd foreground color expression.
+
+@item m3
+Set 3rd frame metadata key from which metadata values will be used to draw a graph.
+
+@item fg3
+Set 3rd foreground color expression.
+
+@item m4
+Set 4th frame metadata key from which metadata values will be used to draw a graph.
+
+@item fg4
+Set 4th foreground color expression.
+
+@item min
+Set minimal value of metadata value.
+
+@item max
+Set maximal value of metadata value.
+
+@item bg
+Set graph background color. Default is white.
+
+@item mode
+Set graph mode.
+
+Available values for mode is:
+@table @samp
+@item bar
+@item dot
+@item line
+@end table
+
+Default is @code{line}.
+
+@item slide
+Set slide mode.
+
+Available values for slide is:
+@table @samp
+@item frame
+Draw new frame when right border is reached.
+
+@item replace
+Replace old columns with new ones.
+
+@item scroll
+Scroll from right to left.
+@end table
+
+Default is @code{frame}.
+
+@item size
+Set size of graph video. For the syntax of this option, check the
+@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}.
+The default value is @code{900x256}.
+
+The foreground color expressions can use the following variables:
+@table @option
+@item MIN
+Minimal value of metadata value.
+
+@item MAX
+Maximal value of metadata value.
+
+@item VAL
+Current metadata key value.
+@end table
+
+The color is defined as 0xAABBGGRR.
+@end table
+
+Example using metadata from @ref{signalstats} filter:
+@example
+signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255
+@end example
+
+Example using metadata from @ref{ebur128} filter:
+@example
+ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5
+@end example
+
@section drawgrid
Draw a grid on the input image.
@end table
+@section erosion
+
+Apply erosion effect to the video.
+
+This filter replaces the pixel by the local(3x3) minimum.
+
+It accepts the following options:
+
+@table @option
+@item threshold0
+@item threshold1
+@item threshold2
+@item threshold3
+Allows to limit the maximum change for each plane, default is 65535.
+If 0, plane will remain unchanged.
+
+@item coordinates
+Flag which specifies the pixel to refer to. Default is 255 i.e. all eight
+pixels are used.
+
+Flags to local 3x3 coordinates maps like this:
+
+ 1 2 3
+ 4 5
+ 6 7 8
+@end table
+
@section extractplanes
Extract color channel components from input video stream into
Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is @code{0}.
@end table
+@section inflate
+
+Apply inflate effect to the video.
+
+This filter replaces the pixel by the local(3x3) average by taking into account
+only values higher than the pixel.
+
+It accepts the following options:
+
+@table @option
+@item threshold0
+@item threshold1
+@item threshold2
+@item threshold3
+Allows to limit the maximum change for each plane, default is 65535.
+If 0, plane will remain unchanged.
+@end table
+
@section interlace
Simple interlacing filter from progressive contents. This interleaves upper (or
@end example
@end itemize
+@section removegrain
+
+The removegrain filter is a spatial denoiser for progressive video.
+
+@table @option
+@item m0
+Set mode for the first plane.
+
+@item m1
+Set mode for the second plane.
+
+@item m2
+Set mode for the third plane.
+
+@item m3
+Set mode for the fourth plane.
+@end table
+
+Range of mode is from 0 to 24. Description of each mode follows:
+
+@table @var
+@item 0
+Leave input plane unchanged. Default.
+
+@item 1
+Clips the pixel with the minimum and maximum of the 8 neighbour pixels.
+
+@item 2
+Clips the pixel with the second minimum and maximum of the 8 neighbour pixels.
+
+@item 3
+Clips the pixel with the third minimum and maximum of the 8 neighbour pixels.
+
+@item 4
+Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels.
+This is equivalent to a median filter.
+
+@item 5
+Line-sensitive clipping giving the minimal change.
+
+@item 6
+Line-sensitive clipping, intermediate.
+
+@item 7
+Line-sensitive clipping, intermediate.
+
+@item 8
+Line-sensitive clipping, intermediate.
+
+@item 9
+Line-sensitive clipping on a line where the neighbours pixels are the closest.
+
+@item 10
+Replaces the target pixel with the closest neighbour.
+
+@item 11
+[1 2 1] horizontal and vertical kernel blur.
+
+@item 12
+Same as mode 11.
+
+@item 13
+Bob mode, interpolates top field from the line where the neighbours
+pixels are the closest.
+
+@item 14
+Bob mode, interpolates bottom field from the line where the neighbours
+pixels are the closest.
+
+@item 15
+Bob mode, interpolates top field. Same as 13 but with a more complicated
+interpolation formula.
+
+@item 16
+Bob mode, interpolates bottom field. Same as 14 but with a more complicated
+interpolation formula.
+
+@item 17
+Clips the pixel with the minimum and maximum of respectively the maximum and
+minimum of each pair of opposite neighbour pixels.
+
+@item 18
+Line-sensitive clipping using opposite neighbours whose greatest distance from
+the current pixel is minimal.
+
+@item 19
+Replaces the pixel with the average of its 8 neighbours.
+
+@item 20
+Averages the 9 pixels ([1 1 1] horizontal and vertical blur).
+
+@item 21
+Clips pixels using the averages of opposite neighbour.
+
+@item 22
+Same as mode 21 but simpler and faster.
+
+@item 23
+Small edge and halo removal, but reputed useless.
+
+@item 24
+Similar as 23.
+@end table
+
@section removelogo
Suppress a TV station logo, using an image file to determine which
This filter uses the repeat_field flag from the Video ES headers and hard repeats
fields based on its value.
+@section reverse
+
+Reverse a clip.
+
+Warning: This iflter qequires memory to buffer the entire clip, so trimming is suggested.
+
+@subsection Examples
+
+@itemize
+@item
+Take the first 5 seconds of a clip, and reverse it.
+@example
+trim=end=5,reverse
+@end example
+@end itemize
+
@section rotate
Rotate video by an arbitrary angle expressed in radians.
ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT
@end example
+@anchor{signalstats}
@section signalstats
Evaluate various visual metrics that assist in determining issues associated
with the digitization of analog video media.
@end itemize
+@anchor{ebur128}
@section ebur128
EBU R128 scanner filter. This filter takes an audio stream as input and outputs
@table @option
@item VOLUME
Current max volume of channel in dB.
+
+@item CHANNEL
+Current channel number, starting from 0.
@end table
@item t
projects / ffmpeg / blobdiff