+atempo=1.25
+@end example
+@end itemize
+
+@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
+ffmpeg -i INPUT -af atrim=60:120
+@end example
+
+@item
+keep only the first 1000 samples
+@example
+ffmpeg -i INPUT -af atrim=end_sample=1000
+@end example
+
+@end itemize
+
+@section bandpass
+
+Apply a two-pole Butterworth band-pass filter with central
+frequency @var{frequency}, and (3dB-point) band-width width.
+The @var{csg} option selects a constant skirt gain (peak gain = Q)
+instead of the default: constant 0dB peak gain.
+The filter roll off at 6dB per octave (20dB per decade).
+
+The filter accepts the following options:
+
+@table @option
+@item frequency, f
+Set the filter's central frequency. Default is @code{3000}.
+
+@item csg
+Constant skirt gain if set to 1. Defaults to 0.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item h
+Hz
+@item q
+Q-Factor
+@item o
+octave
+@item s
+slope
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+@end table
+
+@section bandreject
+
+Apply a two-pole Butterworth band-reject filter with central
+frequency @var{frequency}, and (3dB-point) band-width @var{width}.
+The filter roll off at 6dB per octave (20dB per decade).
+
+The filter accepts the following options:
+
+@table @option
+@item frequency, f
+Set the filter's central frequency. Default is @code{3000}.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item h
+Hz
+@item q
+Q-Factor
+@item o
+octave
+@item s
+slope
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+@end table
+
+@section bass
+
+Boost or cut the bass (lower) frequencies of the audio using a two-pole
+shelving filter with a response similar to that of a standard
+hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
+
+The filter accepts the following options:
+
+@table @option
+@item gain, g
+Give the gain at 0 Hz. Its useful range is about -20
+(for a large cut) to +20 (for a large boost).
+Beware of clipping when using a positive gain.
+
+@item frequency, f
+Set the filter's central frequency and so can be used
+to extend or reduce the frequency range to be boosted or cut.
+The default value is @code{100} Hz.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item h
+Hz
+@item q
+Q-Factor
+@item o
+octave
+@item s
+slope
+@end table
+
+@item width, w
+Determine how steep is the filter's shelf transition.
+@end table
+
+@section biquad
+
+Apply a biquad IIR filter with the given coefficients.
+Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
+are the numerator and denominator coefficients respectively.
+
+@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
+ffmpeg -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
+ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav
+@end example
+
+@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
+ffmpeg -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
+ffmpeg -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 earwax
+
+Make audio easier to listen to on headphones.
+
+This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
+so that when listened to on headphones the stereo image is moved from
+inside your head (standard for headphones) to outside and in front of
+the listener (standard for speakers).
+
+Ported from SoX.
+
+@section equalizer
+
+Apply a two-pole peaking equalisation (EQ) filter. With this
+filter, the signal-level at and around a selected frequency can
+be increased or decreased, whilst (unlike bandpass and bandreject
+filters) that at all other frequencies is unchanged.
+
+In order to produce complex equalisation curves, this filter can
+be given several times, each with a different central frequency.
+
+The filter accepts the following options:
+
+@table @option
+@item frequency, f
+Set the filter's central frequency in Hz.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item h
+Hz
+@item q
+Q-Factor
+@item o
+octave
+@item s
+slope
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+
+@item gain, g
+Set the required gain or attenuation in dB.
+Beware of clipping when using a positive gain.
+@end table
+
+@section highpass
+
+Apply a high-pass filter with 3dB point frequency.
+The filter can be either single-pole, or double-pole (the default).
+The filter roll off at 6dB per pole per octave (20dB per pole per decade).
+
+The filter accepts the following options:
+
+@table @option
+@item frequency, f
+Set frequency in Hz. Default is 3000.
+
+@item poles, p
+Set number of poles. Default is 2.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item h
+Hz
+@item q
+Q-Factor
+@item o
+octave
+@item s
+slope
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+Applies only to double-pole filter.
+The default is 0.707q and gives a Butterworth response.
+@end table
+
+@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
+ffmpeg -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
+ffmpeg -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
projects / ffmpeg.git / commitdiff