+@section asetpts
+
+Change the PTS (presentation timestamp) of the input audio frames.
+
+This filter accepts the following options:
+
+@table @option
+
+@item expr
+The expression which is evaluated for each frame to construct its timestamp.
+
+@end table
+
+The expression is evaluated through the eval API and can contain the following
+constants:
+
+@table @option
+@item PTS
+the presentation timestamp in input
+
+@item PI
+Greek PI
+
+@item PHI
+golden ratio
+
+@item E
+Euler number
+
+@item N
+Number of the audio samples pass through the filter so far, starting at 0.
+
+@item S
+Number of the audio samples in the current frame.
+
+@item SR
+Audio sample rate.
+
+@item STARTPTS
+the PTS of the first frame
+
+@item PREV_INPTS
+previous input PTS
+
+@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:
+
+@example
+# start counting PTS from zero
+asetpts=expr=PTS-STARTPTS
+
+#generate timestamps by counting samples
+asetpts=expr=N/SR/TB
+
+# generate timestamps from a "live source" and rebase onto the current timebase
+asetpts='(RTCTIME - RTCSTART) / (TB * 1000000)"
+@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, 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 atrim
+Trim the input so that the output contains one continuous subpart of the input.
+
+This filter accepts the following options:
+@table @option
+@item start
+Timestamp (in seconds) of the start of the kept section. I.e. the audio sample
+with the timestamp @var{start} will be the first sample in the output.
+
+@item end
+Timestamp (in seconds) of the first audio sample that will be dropped. I.e. the
+audio sample immediately preceding the one with the timestamp @var{end} will be
+the last sample in the output.
+
+@item start_pts
+Same as @var{start}, except this option sets the start timestamp in samples
+instead of seconds.
+
+@item end_pts
+Same as @var{end}, except this option sets the end timestamp in samples instead
+of seconds.
+
+@item duration
+Maximum duration of the output in seconds.
+
+@item start_sample
+Number of the first sample that should be passed to output.
+
+@item end_sample
+Number of the first sample that should be dropped.
+@end table
+
+Note that the first two sets of the start/end options and the @option{duration}
+option look at the frame timestamp, while the _sample options simply count the
+samples that pass through the filter. So start/end_pts and start/end_sample will
+give different results when the timestamps are wrong, inexact or do not start at
+zero. Also note that this filter does not modify the timestamps. If you wish
+that the output timestamps start at zero, insert the asetpts filter after the
+atrim filter.
+
+If multiple start or end options are set, this filter tries to be greedy and
+keep all samples that match at least one of the specified constraints. To keep
+only the part that matches all the constraints at once, chain multiple atrim
+filters.
+
+The defaults are such that all the input is kept. So it is possible to set e.g.
+just the end values to keep everything before the specified time.
+
+Examples:
+@itemize
+@item
+drop everything except the second minute of input
+@example
+avconv -i INPUT -af atrim=60:120
+@end example
+
+@item
+keep only the first 1000 samples
+@example
+avconv -i INPUT -af atrim=end_sample=1000
+@end example
+
+@end itemize
+
+@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 '|'-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 compand
+Compress or expand audio dynamic range.
+
+A description of the accepted options follows.
+
+@table @option
+
+@item attacks
+@item decays
+Set list of times in seconds for each channel over which the instantaneous level
+of the input signal is averaged to determine its volume. @var{attacks} refers to
+increase of volume and @var{decays} refers to decrease of volume. For most
+situations, the attack time (response to the audio getting louder) should be
+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.
+
+@item points
+Set list of points for the transfer function, specified in dB relative to the
+maximum possible signal amplitude. Each key points list must be defined using
+the following syntax: @code{x0/y0|x1/y1|x2/y2|....}
+
+The input values must be in strictly increasing order but the transfer function
+does not have to be monotonically rising. The point @code{0/0} is assumed but
+may be overridden (by @code{0/out-dBn}). Typical values for the transfer
+function are @code{-70/-70|-60/-20}.
+
+@item soft-knee
+Set the curve radius in dB for all joints. Defaults to 0.01.
+
+@item gain
+Set additional gain in dB to be applied at all points on the transfer function.
+This allows easy adjustment of the overall gain. Defaults to 0.
+
+@item volume
+Set initial volume in dB to be assumed for each channel when filtering starts.
+This permits the user to supply a nominal level initially, so that, for
+example, a very large gain is not applied to initial signal levels before the
+companding has begun to operate. A typical value for audio which is initially
+quiet is -90 dB. Defaults to 0.
+
+@item delay
+Set delay in seconds. The input audio is analyzed immediately, but audio is
+delayed before being fed to the volume adjuster. Specifying a delay
+approximately equal to the attack/decay times allows the filter to effectively
+operate in predictive rather than reactive mode. Defaults to 0.
+
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Make music with both quiet and loud passages suitable for listening in a noisy
+environment:
+@example
+compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
+@end example
+
+@item
+Noise gate for when the noise is at a lower level than the signal:
+@example
+compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
+@end example
+
+@item
+Here is another noise gate, this time for when the noise is at a higher level
+than the signal (making it, in some ways, similar to squelch):
+@example
+compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
+@end example
+@end itemize
+
+@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 '|'-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
+