@section aresample
-Resample the input audio to the specified sample rate.
+Resample the input audio to the specified parameters. If none are specified
+then the filter will automatically convert between its input
+and output.
-The filter accepts exactly one parameter, the output sample rate. If not
-specified then the filter will automatically convert between its input
-and output sample rates.
+This filter is also able to stretch/squeeze the audio data to make it match
+the timestamps or to inject silence / cut out audio to make it match the
+timestamps, do a combination of both or do neither.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item min_comp
+Minimum difference between timestamps and audio data (in seconds) to trigger
+stretching/squeezing/filling or trimming of the data to make it match the
+timestamps. The default is that stretching/squeezing/filling and
+trimming is disabled (min_comp = infinite).
+
+@item min_hard_comp
+Minimum difference between timestamps and audio data (in seconds) to trigger
+adding/dropping samples to make it match the timestamps.
+This option effectively is a threshold to select between hard (trim/fill) and
+soft (squeeze/stretch) compensation. Note that all compensation is by default
+disabled through min_comp.
+The default is 0.1 seconds.
+
+@item max_soft_comp
+Maximum stretch/squeeze factor.
+Default value 0.
+
+@item tsf, internal_sample_fmt
+Internal sampling format.
+Default is automatic selection
+
+@item clev, center_mix_level
+center mix level, for rematrixing
+Default is 3.0dB
+
+@item slev, surround_mix_level
+surround mix level, for rematrixing
+Default is 3.0dB
+
+@item rmvol, rematrix_volume
+rematrix volume
+Default is 1.0
+
+@item lfe_mix_level
+Low frequency effects mix level.
+Default is 0
+
+@item matrix_encoding
+matrixed stereo encoding
+@table @option
+@item none
+No matrixed stereo encoding
+
+@item dolby
+Dolby matrixed stereo encoding
+
+@item dolby
+Dolby Pro Logic II matrixed stereo encoding
+@end table
+
+Default value is @code{none}.
+
+@end table
For example, to resample the input audio to 44100Hz:
@example
ffmpeg -f lavfi -i amovie=silence.mp3,silencedetect=noise=0.0001 -f null -
@end example
-@section volume
-
-Adjust the input audio volume.
-
-The filter accepts exactly one parameter @var{vol}, which expresses
-how the audio volume will be increased or decreased.
-
-Output values are clipped to the maximum value.
-
-If @var{vol} is expressed as a decimal number, the output audio
-volume is given by the relation:
-@example
-@var{output_volume} = @var{vol} * @var{input_volume}
-@end example
-
-If @var{vol} is expressed as a decimal number followed by the string
-"dB", the value represents the requested change in decibels of the
-input audio power, and the output audio volume is given by the
-relation:
-@example
-@var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
-@end example
-
-Otherwise @var{vol} is considered an expression and its evaluated
-value is used for computing the output audio volume according to the
-first relation.
-
-Default value for @var{vol} is 1.0.
-
-@subsection Examples
-
-@itemize
-@item
-Half the input audio volume:
-@example
-volume=0.5
-@end example
-
-The above example is equivalent to:
-@example
-volume=1/2
-@end example
-
-@item
-Decrease input audio power by 12 decibels:
-@example
-volume=-12dB
-@end example
-@end itemize
-
-@section volumedetect
-
-Detect the volume of the input video.
-
-The filter has no parameters. The input is not modified. Statistics about
-the volume will be printed in the log when the input stream end is reached.
-
-In particular it will show the mean volume (root mean square), maximum
-volume (on a per-sample basis), and the beginning of an histogram of the
-registered volume values (from the maximum value to a cumulated 1/1000 of
-the samples).
-
-All volumes are in decibels relative to the maximum PCM value.
-
-Here is an excerpt of the output:
-@example
-[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB
-[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB
-[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6
-[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62
-[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286
-[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042
-[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551
-[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609
-[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409
-@end example
-
-It means that:
-@itemize
-@item
-The mean square energy is approximately -27 dB, or 10^-2.7.
-@item
-The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
-@item
-There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
-@end itemize
-
-In other words, raising the volume by +4 dB does not cause any clipping,
-raising it by +5 dB causes clipping for 6 samples, etc.
-
@section asyncts
Synchronize audio data with timestamps by squeezing/stretching it and/or
dropping samples/adding silence when needed.
Convert the audio sample format, sample rate and channel layout. This filter is
not meant to be used directly.
+@section volume
+
+Adjust the input audio volume.
+
+The filter accepts the following named parameters. If the key of the
+first options is omitted, the arguments are interpreted according to
+the following syntax:
+@example
+volume=@var{volume}:@var{precision}
+@end example
+
+@table @option
+
+@item volume
+Expresses how the audio volume will be increased or decreased.
+
+Output values are clipped to the maximum value.
+
+The output audio volume is given by the relation:
+@example
+@var{output_volume} = @var{volume} * @var{input_volume}
+@end example
+
+Default value for @var{volume} is 1.0.
+
+@item precision
+Set the mathematical precision.
+
+This determines which input sample formats will be allowed, which affects the
+precision of the volume scaling.
+
+@table @option
+@item fixed
+8-bit fixed-point; limits input sample format to U8, S16, and S32.
+@item float
+32-bit floating-point; limits input sample format to FLT. (default)
+@item double
+64-bit floating-point; limits input sample format to DBL.
+@end table
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Halve the input audio volume:
+@example
+volume=volume=0.5
+volume=volume=1/2
+volume=volume=-6.0206dB
+@end example
+
+In all the above example the named key for @option{volume} can be
+omitted, for example like in:
+@example
+volume=0.5
+@end example
+
+@item
+Increase input audio power by 6 decibels using fixed-point precision:
+@example
+volume=volume=6dB:precision=fixed
+@end example
+@end itemize
+
+@section volumedetect
+
+Detect the volume of the input video.
+
+The filter has no parameters. The input is not modified. Statistics about
+the volume will be printed in the log when the input stream end is reached.
+
+In particular it will show the mean volume (root mean square), maximum
+volume (on a per-sample basis), and the beginning of an histogram of the
+registered volume values (from the maximum value to a cumulated 1/1000 of
+the samples).
+
+All volumes are in decibels relative to the maximum PCM value.
+
+Here is an excerpt of the output:
+@example
+[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB
+[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB
+[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6
+[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62
+[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286
+[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042
+[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551
+[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609
+[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409
+@end example
+
+It means that:
+@itemize
+@item
+The mean square energy is approximately -27 dB, or 10^-2.7.
+@item
+The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
+@item
+There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
+@end itemize
+
+In other words, raising the volume by +4 dB does not cause any clipping,
+raising it by +5 dB causes clipping for 6 samples, etc.
+
@c man end AUDIO FILTERS
@chapter Audio Sources
@section ass
-Draw ASS (Advanced Substation Alpha) subtitles on top of input video
-using the libass library.
-
-To enable compilation of this filter you need to configure FFmpeg with
-@code{--enable-libass}.
-
-This filter accepts the following named options, expressed as a
-sequence of @var{key}=@var{value} pairs, separated by ":".
-
-@table @option
-@item filename, f
-Set the filename of the ASS file to read. It must be specified.
-
-@item original_size
-Specify the size of the original video, the video for which the ASS file
-was composed. Due to a misdesign in ASS aspect ratio arithmetic, this is
-necessary to correctly scale the fonts if the aspect ratio has been changed.
-@end table
-
-If the first key is not specified, it is assumed that the first value
-specifies the @option{filename}.
-
-For example, to render the file @file{sub.ass} on top of the input
-video, use the command:
-@example
-ass=sub.ass
-@end example
-
-which is equivalent to:
-@example
-ass=filename=sub.ass
-@end example
+Same as the @ref{subtitles} filter, except that it doesn't require libavcodec
+and libavformat to work. On the other hand, it is limited to ASS (Advanced
+Substation Alpha) subtitles files.
@section bbox
If both @var{text} and @var{textfile} are specified, an error is thrown.
+@item reload
+If set to 1, the @var{textfile} will be reloaded before each frame.
+Be sure to update it atomically, or it may be read partially, or even fail.
+
@item x, y
The expressions which specify the offsets where text will be drawn
within the video frame. They are relative to the top/left border of the
@table @option
@item fps
-Desired output framerate.
+Desired output framerate. The default is @code{25}.
@item round
-Rounding method. The default is @code{near}.
+Rounding method.
+Possible values are:
+@table @option
+@item zero
+zero round towards 0
+@item inf
+round away from 0
+@item down
+round towards -infinity
+@item up
+round towards +infinity
+@item near
+round to nearest
@end table
+The default is @code{near}.
+
+@end table
+
+Alternatively, the options can be specified as a flat string:
+@var{fps}[:@var{round}].
+
+See also the @ref{setpts} filter.
@section framestep
plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and
@code{0.5,0.5} for chroma planes.
+@item T
+Time of the current frame, expressed in seconds.
+
@item p(x, y)
Return the value of the pixel at location (@var{x},@var{y}) of the current
plane.
geq=p(W-X\,Y)
@end example
+@item
+Generate a bidimensional sine wave, with angle @code{PI/3} and a
+wavelength of 100 pixels:
+@example
+geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128
+@end example
+
@item
Generate a fancy enigmatic moving light:
@example
The list of the currently supported filters follows:
@table @var
-@item denoise3d
@item detc
@item dint
@item divtc
corresponding property, which affects how the frame is treated by
following filters (e.g. @code{fieldorder} or @code{yadif}).
-It accepts a string parameter, which can assume the following values:
+This filter accepts a single option @option{mode}, which can be
+specified either by setting @code{mode=VALUE} either setting the
+value alone. Available values are:
+
@table @samp
@item auto
Keep the same field property.
image, a value included in [0,30] will filter flat areas and a value
included in [-30,0] will filter edges.
+@anchor{subtitles}
+@section subtitles
+
+Draw subtitles on top of input video using the libass library.
+
+To enable compilation of this filter you need to configure FFmpeg with
+@code{--enable-libass}. This filter also requires a build with libavcodec and
+libavformat to convert the passed subtitles file to ASS (Advanced Substation
+Alpha) subtitles format.
+
+This filter accepts the following named options, expressed as a
+sequence of @var{key}=@var{value} pairs, separated by ":".
+
+@table @option
+@item filename, f
+Set the filename of the subtitle file to read. It must be specified.
+
+@item original_size
+Specify the size of the original video, the video for which the ASS file
+was composed. Due to a misdesign in ASS aspect ratio arithmetic, this is
+necessary to correctly scale the fonts if the aspect ratio has been changed.
+@end table
+
+If the first key is not specified, it is assumed that the first value
+specifies the @option{filename}.
+
+For example, to render the file @file{sub.srt} on top of the input
+video, use the command:
+@example
+subtitles=sub.srt
+@end example
+
+which is equivalent to:
+@example
+subtitles=filename=sub.srt
+@end example
+
@section split
Split input video into several identical outputs.
Frames are counted starting from 1, so the first input frame is
considered odd.
-This filter accepts a single parameter specifying the mode. Available
-modes are:
+This filter accepts a single option @option{mode} specifying the mode,
+which can be specified either by specyfing @code{mode=VALUE} either
+specifying the value alone. Available values are:
@table @samp
@item merge, 0
@end example
@end itemize
+@anchor{setpts}
@section asetpts, setpts
Change the PTS (presentation timestamp) of the input frames.
projects / ffmpeg / blobdiff