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 the following named parameters:
+@table @option
+
+@item sample_fmts
+A comma-separated list of requested sample formats.
+
+@item sample_rates
+A comma-separated list of requested sample rates.
+
+@item channel_layouts
+A comma-separated list of requested channel layouts.
+
+@end table
+
+If a parameter is omitted, all values are allowed.
+
+For example to force the output to either unsigned 8-bit or signed 16-bit stereo:
+@example
+aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo
+@end example
+
+@section amix
+
+Mixes multiple audio inputs into a single output.
+
+For example
+@example
+avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
+@end example
+will mix 3 input audio streams to a single output with the same duration as the
+first input and a dropout transition time of 3 seconds.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item inputs
+Number of inputs. If unspecified, it defaults to 2.
+
+@item duration
+How to determine the end-of-stream.
+@table @option
+
+@item longest
+Duration of longest input. (default)
+
+@item shortest
+Duration of shortest input.
+
+@item first
+Duration of first input.
+
+@end table
+
+@item dropout_transition
+Transition time, in seconds, for volume renormalization when an input
+stream ends. The default value is 2 seconds.
+
+@end table
+
@section anull
Pass the audio source unchanged to the output.
+@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, in time base units; the time base
+depends on the filter input pad, and is usually 1/@var{sample_rate}.
+
+@item pts_time
+presentation timestamp of the input frame in seconds
+
+@item fmt
+sample format
+
+@item chlayout
+channel layout
+
+@item rate
+sample rate for the audio frame
+
+@item nb_samples
+number of samples (per channel) in the frame
+
+@item checksum
+Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio
+the data is treated as if all the planes were concatenated.
+
+@item plane_checksums
+A list of Adler-32 checksums for each data plane.
+@end table
+
+@section asplit
+
+Split input audio into several identical outputs.
+
+The filter accepts a single parameter which specifies the number of outputs. If
+unspecified, it defaults to 2.
+
+For example
+@example
+avconv -i INPUT -filter_complex asplit=5 OUTPUT
+@end example
+will create 5 copies of the input audio.
+
+@section asyncts
+Synchronize audio data with timestamps by squeezing/stretching it and/or
+dropping samples/adding silence when needed.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item compensate
+Enable stretching/squeezing the data to make it match the timestamps. Disabled
+by default. When disabled, time gaps are covered with silence.
+
+@item min_delta
+Minimum difference between timestamps and audio data (in seconds) to trigger
+adding/dropping samples. Default value is 0.1. If you get non-perfect sync with
+this filter, try setting this parameter to 0.
+
+@item max_comp
+Maximum compensation in samples per second. Relevant only with compensate=1.
+Default value 500.
+
+@item first_pts
+Assume the first pts should be this value. The time base is 1 / sample rate.
+This allows for padding/trimming at the start of stream. By default, no
+assumption is made about the first frame's expected pts, so no padding or
+trimming is done. For example, this could be set to 0 to pad the beginning with
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
+
+@end table
+
+@section channelsplit
+Split each channel in input audio stream into a separate output stream.
+
+This filter accepts the following named parameters:
+@table @option
+@item channel_layout
+Channel layout of the input stream. Default is "stereo".
+@end table
+
+For example, assuming a stereo input MP3 file
+@example
+avconv -i in.mp3 -filter_complex channelsplit out.mkv
+@end example
+will create an output Matroska file with two audio streams, one containing only
+the left channel and the other the right channel.
+
+To split a 5.1 WAV file into per-channel files
+@example
+avconv -i in.wav -filter_complex
+'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
+-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
+front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
+side_right.wav
+@end example
+
+@section channelmap
+Remap input channels to new locations.
+
+This filter accepts the following named parameters:
+@table @option
+@item channel_layout
+Channel layout of the output stream.
+
+@item map
+Map channels from input to output. The argument is a comma-separated list of
+mappings, each in the @code{@var{in_channel}-@var{out_channel}} or
+@var{in_channel} form. @var{in_channel} can be either the name of the input
+channel (e.g. FL for front left) or its index in the input channel layout.
+@var{out_channel} is the name of the output channel or its index in the output
+channel layout. If @var{out_channel} is not given then it is implicitly an
+index, starting with zero and increasing by one for each mapping.
+@end table
+
+If no mapping is present, the filter will implicitly map input channels to
+output channels preserving index.
+
+For example, assuming a 5.1+downmix input MOV file
+@example
+avconv -i in.mov -filter 'channelmap=map=DL-FL\,DR-FR' out.wav
+@end example
+will create an output WAV file tagged as stereo from the downmix channels of
+the input.
+
+To fix a 5.1 WAV improperly encoded in AAC's native channel order
+@example
+avconv -i in.wav -filter 'channelmap=1\,2\,0\,5\,3\,4:channel_layout=5.1' out.wav
+@end example
+
+@section join
+Join multiple input streams into one multi-channel stream.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item inputs
+Number of input streams. Defaults to 2.
+
+@item channel_layout
+Desired output channel layout. Defaults to stereo.
+
+@item map
+Map channels from inputs to output. The argument is a comma-separated list of
+mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}}
+form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel}
+can be either the name of the input channel (e.g. FL for front left) or its
+index in the specified input stream. @var{out_channel} is the name of the output
+channel.
+@end table
+
+The filter will attempt to guess the mappings when those are not specified
+explicitly. It does so by first trying to find an unused matching input channel
+and if that fails it picks the first unused input channel.
+
+E.g. to join 3 inputs (with properly set channel layouts)
+@example
+avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
+@end example
+
+To build a 5.1 output from 6 single-channel streams:
+@example
+avconv -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
+'join=inputs=6:channel_layout=5.1:map=0.0-FL\,1.0-FR\,2.0-FC\,3.0-SL\,4.0-SR\,5.0-LFE'
+out
+@end example
+
+@section resample
+Convert the audio sample format, sample rate and channel layout. This filter is
+not meant to be used directly, it is inserted automatically by libavfilter
+whenever conversion is needed. Use the @var{aformat} filter to force a specific
+conversion.
+
+@section volume
+
+Adjust the input audio volume.
+
+The filter accepts the following named parameters:
+@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
+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
+
+@item
+Increase input audio power by 6 decibels using fixed-point precision:
+@example
+volume=volume=6dB:precision=fixed
+@end example
+@end itemize
+
@c man end AUDIO FILTERS
@chapter Audio Sources
of @var{channel_layout} is 3, which corresponds to CH_LAYOUT_STEREO.
Check the channel_layout_map definition in
-@file{libavcodec/audioconvert.c} for the mapping between strings and
+@file{libavutil/channel_layout.c} for the mapping between strings and
channel layout values.
Follow some examples:
anullsrc=48000:mono
@end example
+@section abuffer
+Buffer audio frames, and make them available to the filter chain.
+
+This source is not intended to be part of user-supplied graph descriptions but
+for insertion by calling programs through the interface defined in
+@file{libavfilter/buffersrc.h}.
+
+It accepts the following named parameters:
+@table @option
+
+@item time_base
+Timebase which will be used for timestamps of submitted frames. It must be
+either a floating-point number or in @var{numerator}/@var{denominator} form.
+
+@item sample_rate
+Audio sample rate.
+
+@item sample_fmt
+Name of the sample format, as returned by @code{av_get_sample_fmt_name()}.
+
+@item channel_layout
+Channel layout of the audio data, in the form that can be accepted by
+@code{av_get_channel_layout()}.
+@end table
+
+All the parameters need to be explicitly defined.
+
@c man end AUDIO SOURCES
@chapter Audio Sinks
mainly useful as a template and to be employed in analysis / debugging
tools.
+@section abuffersink
+This sink is intended for programmatic use. Frames that arrive on this sink can
+be retrieved by the calling program using the interface defined in
+@file{libavfilter/buffersink.h}.
+
+This filter accepts no parameters.
+
@c man end AUDIO SINKS
@chapter Video Filters
format=yuv420p:yuv444p:yuv410p
@end example
+@section fps
+
+Convert the video to specified constant framerate by duplicating or dropping
+frames as necessary.
+
+This filter accepts the following named parameters:
+@table @option
+
+@item fps
+Desired output framerate.
+
+@end table
+
@anchor{frei0r}
@section frei0r
The @var{lut} filter requires either YUV or RGB pixel formats in
input, and accepts the options:
@table @option
-@var{c0} (first pixel component)
-@var{c1} (second pixel component)
-@var{c2} (third pixel component)
-@var{c3} (fourth pixel component, corresponds to the alpha component)
+@item @var{c0} (first pixel component)
+@item @var{c1} (second pixel component)
+@item @var{c2} (third pixel component)
+@item @var{c3} (fourth pixel component, corresponds to the alpha component)
@end table
The exact component associated to each option depends on the format in
The @var{lutrgb} filter requires RGB pixel formats in input, and
accepts the options:
@table @option
-@var{r} (red component)
-@var{g} (green component)
-@var{b} (blue component)
-@var{a} (alpha component)
+@item @var{r} (red component)
+@item @var{g} (green component)
+@item @var{b} (blue component)
+@item @var{a} (alpha component)
@end table
The @var{lutyuv} filter requires YUV pixel formats in input, and
accepts the options:
@table @option
-@var{y} (Y/luminance component)
-@var{u} (U/Cb component)
-@var{v} (V/Cr component)
-@var{a} (alpha component)
+@item @var{y} (Y/luminance component)
+@item @var{u} (U/Cb component)
+@item @var{v} (V/Cr component)
+@item @var{a} (alpha component)
@end table
The expressions can contain the following constants and functions:
@item PREV_OUTPTS
previous output PTS
+@item RTCTIME
+wallclock (RTC) time in microseconds
+
+@item RTCSTART
+wallclock (RTC) time at the start of the movie in microseconds
+
@end table
Some examples follow:
# fixed rate 25 fps with some jitter
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
+
+# generate timestamps from a "live source" and rebase onto the current timebase
+setpts='(RTCTIME - RTCSTART) / (TB * 1000000)"
@end example
@anchor{setsar}
"[@var{c0} @var{c1} @var{c2} @var{c3}]"
@end table
-@section slicify
-
-Pass the images of input video on to next video filter as multiple
-slices.
-
-@example
-./avconv -i in.avi -vf "slicify=32" out.avi
-@end example
-
-The filter accepts the slice height as parameter. If the parameter is
-not specified it will use the default value of 16.
-
-Adding this in the beginning of filter chains should make filtering
-faster due to better use of the memory cache.
-
@section split
Split input video into several identical outputs.
with format "yuv410p", assuming 1/24 as the timestamps timebase and
square pixels (1:1 sample aspect ratio).
Since the pixel format with name "yuv410p" corresponds to the number 6
-(check the enum PixelFormat definition in @file{libavutil/pixfmt.h}),
+(check the enum AVPixelFormat definition in @file{libavutil/pixfmt.h}),
this example corresponds to:
@example
buffer=320:240:6:1:24
Read a video stream from a movie container.
+Note that this source is a hack that bypasses the standard input path. It can be
+useful in applications that do not support arbitrary filter graphs, but its use
+is discouraged in those that do. Specifically in @command{avconv} this filter
+should never be used, the @option{-filter_complex} option fully replaces it.
+
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),
Below is a description of the currently available video sinks.
+@section buffersink
+
+Buffer video frames, and make them available to the end of the filter
+graph.
+
+This sink is intended for a programmatic use through the interface defined in
+@file{libavfilter/buffersink.h}.
+
@section nullsink
Null video sink, do absolutely nothing with the input video. It is
projects / ffmpeg / blobdiff