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
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/asink_abuffer.h}.
+through the interface defined in @file{libavfilter/buffersink.h}.
-It requires a pointer to a ABufferSinkContext structure, which defines the
-incoming buffers' format, to be passed as the opaque parameter to
-@code{avfilter_init_filter} for initialization.
+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
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
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