A filtergraph can be represented using a textual representation, which
is recognized by the @code{-vf} and @code{-af} options of the ff*
-tools, and by the @code{av_parse_graph()} function defined in
-@file{libavfilter/avfiltergraph}.
+tools, and by the @code{avfilter_graph_parse()} function defined in
+@file{libavfilter/avfiltergraph.h}.
A filterchain consists of a sequence of connected filters, each one
connected to the previous one in the sequence. A filterchain is
Below is a description of the currently available audio filters.
+@section aformat
+
+Convert the input audio to one of the specified formats. The framework will
+negotiate the most appropriate format to minimize conversions.
+
+The filter accepts three lists of formats, separated by ":", in the form:
+"@var{sample_formats}:@var{channel_layouts}:@var{packing_formats}".
+
+Elements in each list are separated by "," which has to be escaped in the
+filtergraph specification.
+
+The special parameter "all", in place of a list of elements, signifies all
+supported formats.
+
+Some examples follow:
+@example
+aformat=u8\\,s16:mono:packed
+
+aformat=s16:mono\\,stereo:all
+@end example
+
@section anull
Pass the audio source unchanged to the output.
+@section aresample
+
+Resample the input audio to the specified sample rate.
+
+The filter accepts exactly one parameter, the output sample rate. If not
+specified then the filter will automatically convert between its input
+and output sample rates.
+
+For example, to resample the input audio to 44100Hz:
+@example
+aresample=44100
+@end example
+
+@section ashowinfo
+
+Show a line containing various information for each input audio frame.
+The input audio is not modified.
+
+The shown line contains a sequence of key/value pairs of the form
+@var{key}:@var{value}.
+
+A description of each shown parameter follows:
+
+@table @option
+@item n
+sequential number of the input frame, starting from 0
+
+@item pts
+presentation TimeStamp of the input frame, expressed as a number of
+time base units. The time base unit depends on the filter input pad, and
+is usually 1/@var{sample_rate}.
+
+@item pts_time
+presentation TimeStamp of the input frame, expressed as a number of
+seconds
+
+@item pos
+position of the frame in the input stream, -1 if this information in
+unavailable and/or meanigless (for example in case of synthetic audio)
+
+@item fmt
+sample format name
+
+@item chlayout
+channel layout description
+
+@item nb_samples
+number of samples (per each channel) contained in the filtered frame
+
+@item rate
+sample rate for the audio frame
+
+@item planar
+if the packing format is planar, 0 if packed
+
+@item checksum
+Adler-32 checksum of all the planes of the input frame
+
+@item plane_checksum
+Adler-32 checksum for each input frame plane, expressed in the form
+"[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5} @var{c6} @var{c7}]"
+@end table
+
@c man end AUDIO FILTERS
@chapter Audio Sources
Below is a description of the currently available audio sources.
+@section abuffer
+
+Buffer audio frames, and make them available to the filter chain.
+
+This source is mainly intended for a programmatic use, in particular
+through the interface defined in @file{libavfilter/asrc_abuffer.h}.
+
+It accepts the following mandatory parameters:
+@var{sample_rate}:@var{sample_fmt}:@var{channel_layout}:@var{packing}
+
+@table @option
+
+@item sample_rate
+The sample rate of the incoming audio buffers.
+
+@item sample_fmt
+The sample format of the incoming audio buffers.
+Either a sample format name or its corresponging integer representation from
+the enum AVSampleFormat in @file{libavutil/samplefmt.h}
+
+@item channel_layout
+The channel layout of the incoming audio buffers.
+Either a channel layout name from channel_layout_map in
+@file{libavutil/audioconvert.c} or its corresponding integer representation
+from the AV_CH_LAYOUT_* macros in @file{libavutil/audioconvert.h}
+
+@item packing
+Either "packed" or "planar", or their integer representation: 0 or 1
+respectively.
+
+@end table
+
+For example:
+@example
+abuffer=44100:s16:stereo:planar
+@end example
+
+will instruct the source to accept planar 16bit signed stereo at 44100Hz.
+Since the sample format with name "s16" corresponds to the number
+1 and the "stereo" channel layout corresponds to the value 3, this is
+equivalent to:
+@example
+abuffer=44100:1:3:1
+@end example
+
+@section amovie
+
+Read an audio stream from a movie container.
+
+It accepts the syntax: @var{movie_name}[:@var{options}] where
+@var{movie_name} is the name of the resource to read (not necessarily
+a file but also a device or a stream accessed through some protocol),
+and @var{options} is an optional sequence of @var{key}=@var{value}
+pairs, separated by ":".
+
+The description of the accepted options follows.
+
+@table @option
+
+@item format_name, f
+Specify the format assumed for the movie to read, and can be either
+the name of a container or an input device. If not specified the
+format is guessed from @var{movie_name} or by probing.
+
+@item seek_point, sp
+Specify the seek point in seconds, the frames will be output
+starting from this seek point, the parameter is evaluated with
+@code{av_strtod} so the numerical value may be suffixed by an IS
+postfix. Default value is "0".
+
+@item stream_index, si
+Specify the index of the audio stream to read. If the value is -1,
+the best suited audio stream will be automatically selected. Default
+value is "-1".
+
+@end table
+
@section anullsrc
Null audio source, never return audio frames. It is mainly useful as a
Below is a description of the currently available audio sinks.
+@section abuffersink
+
+Buffer audio frames, and make them available to the end of filter chain.
+
+This sink is mainly intended for programmatic use, in particular
+through the interface defined in @file{libavfilter/buffersink.h}.
+
+It requires a pointer to an AVABufferSinkContext structure, which
+defines the incoming buffers' formats, to be passed as the opaque
+parameter to @code{avfilter_init_filter} for initialization.
+
@section anullsink
Null audio sink, do absolutely nothing with the input audio. It is
Apply boxblur algorithm to the input video.
This filter accepts the parameters:
-@var{luma_power}:@var{luma_radius}:@var{chroma_radius}:@var{chroma_power}:@var{alpha_radius}:@var{alpha_power}
+@var{luma_radius}:@var{luma_power}:@var{chroma_radius}:@var{chroma_power}:@var{alpha_radius}:@var{alpha_power}
Chroma and alpha parameters are optional, if not specified they default
to the corresponding values set for @var{luma_radius} and
playback.
@end table
+@section delogo
+
+Suppress a TV station logo by a simple interpolation of the surrounding
+pixels. Just set a rectangle covering the logo and watch it disappear
+(and sometimes something even uglier appear - your mileage may vary).
+
+The filter accepts parameters as a string of the form
+"@var{x}:@var{y}:@var{w}:@var{h}:@var{band}", or as a list of
+@var{key}=@var{value} pairs, separated by ":".
+
+The description of the accepted parameters follows.
+
+@table @option
+
+@item x, y
+Specify the top left corner coordinates of the logo. They must be
+specified.
+
+@item w, h
+Specify the width and height of the logo to clear. They must be
+specified.
+
+@item band, t
+Specify the thickness of the fuzzy edge of the rectangle (added to
+@var{w} and @var{h}). The default value is 4.
+
+@item show
+When set to 1, a green rectangle is drawn on the screen to simplify
+finding the right @var{x}, @var{y}, @var{w}, @var{h} parameters, and
+@var{band} is set to 4. The default value is 0.
+
+@end table
+
+Some examples follow.
+
+@itemize
+
+@item
+Set a rectangle covering the area with top left corner coordinates 0,0
+and size 100x77, setting a band of size 10:
+@example
+delogo=0:0:100:77:10
+@end example
+
+@item
+As the previous example, but use named options:
+@example
+delogo=x=0:y=0:w=100:h=77:band=10
+@end example
+
+@end itemize
+
@section drawbox
Draw a colored box on the input image.
@table @var
@item 2xsai
@item decimate
-@item delogo
@item denoise3d
@item detc
@item dint
@item framestep
@item fspp
@item geq
-@item gradfun
@item harddup
@item hqdn3d
@item hue
@item spp
@item swapuv
@item telecine
-@item test
@item tile
@item tinterlace
@item unsharp
# pad the input to get a final w/h ratio of 16:9
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
+# for anamorphic video, in order to set the output display aspect ratio,
+# it is necessary to use sar in the expression, according to the relation:
+# (ih * X / ih) * sar = output_dar
+# X = output_dar / sar
+pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"
+
# double output size and put the input video in the bottom-right
# corner of the output padded area
pad="2*iw:2*ih:ow-iw:oh-ih"
Negative values for the amount will blur the input video, while positive
values will sharpen. All parameters are optional and default to the
-equivalent of the string '5:5:1.0:0:0:0.0'.
+equivalent of the string '5:5:1.0:5:5:0.0'.
@table @option
@item chroma_msize_x
Set the chroma matrix horizontal size. It can be an integer between 3
-and 13, default value is 0.
+and 13, default value is 5.
@item chroma_msize_y
Set the chroma matrix vertical size. It can be an integer between 3
-and 13, default value is 0.
+and 13, default value is 5.
-@item luma_amount
+@item chroma_amount
Set the chroma effect strength. It can be a float number between -2.0
and 5.0, default value is 0.0.
@table @option
@item 0
-assume bottom field first
-@item 1
assume top field first
+@item 1
+assume bottom field first
@item -1
enable automatic detection
@end table
@end example
+@section mptestsrc
+
+Generate various test patterns, as generated by the MPlayer test filter.
+
+The size of the generated video is fixed, and is 256x256.
+This source is useful in particular for testing encoding features.
+
+This source accepts an optional sequence of @var{key}=@var{value} pairs,
+separated by ":". The description of the accepted options follows.
+
+@table @option
+
+@item rate, r
+Specify the frame rate of the sourced video, as the number of frames
+generated per second. It has to be a string in the format
+@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
+number or a valid video frame rate abbreviation. The default value is
+"25".
+
+@item duration, d
+Set the video duration of the sourced video. The accepted syntax is:
+@example
+[-]HH[:MM[:SS[.m...]]]
+[-]S+[.m...]
+@end example
+See also the function @code{av_parse_time()}.
+
+If not specified, or the expressed duration is negative, the video is
+supposed to be generated forever.
+
+@item test, t
+
+Set the number or the name of the test to perform. Supported tests are:
+@table @option
+@item dc_luma
+@item dc_chroma
+@item freq_luma
+@item freq_chroma
+@item amp_luma
+@item amp_chroma
+@item cbp
+@item mv
+@item ring1
+@item ring2
+@item all
+@end table
+
+Default value is "all", which will cycle through the list of all tests.
+@end table
+
+For example the following:
+@example
+testsrc=t=dc_luma
+@end example
+
+will generate a "dc_luma" test pattern.
+
@section nullsrc
Null video source, never return images. It is mainly useful as a
number or a valid video frame rate abbreviation. The default value is
"25".
+@item sar
+Set the sample aspect ratio of the sourced video.
+
@item duration
Set the video duration of the sourced video. The accepted syntax is:
@example
graph.
This sink is mainly intended for a programmatic use, in particular
-through the interface defined in @file{libavfilter/vsink_buffer.h}.
+through the interface defined in @file{libavfilter/buffersink.h}.
It does not require a string parameter in input, but you need to
specify a pointer to a list of supported pixel formats terminated by
projects / ffmpeg / blobdiff