doc/filters: sort audio filters by name

2013-05-10 13:26:12 +02:00 · 2013-05-10 13:26:12 +02:00 · edc05698aa
commit edc05698aa
parent a9705e4de9
1 changed files with 468 additions and 462 deletions
--- a/doc/filters.texi
+++ b/doc/filters.texi
@ -347,282 +347,6 @@ aconvert=u8:auto
@end example
@end itemize
@section allpass
 Apply a two-pole all-pass filter with central frequency (in Hz)
@var{frequency}, and filter-width @var{width}.
 An all-pass filter changes the audio's frequency to phase relationship
 without changing its frequency to amplitude relationship.
 The filter accepts the following options:
@table @option
@item frequency, f
 Set 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.
@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 lowpass
 Apply a low-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 500.
@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 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 treble
 Boost or cut treble (upper) 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 whichever is the lower of ~22 kHz and the
 Nyquist frequency. 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{3000} 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 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 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 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 afade
 Apply fade-in/out effect to input audio.
@ -738,6 +462,36 @@ For example to force the output to either unsigned 8-bit or signed 16-bit stereo
 aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
@section allpass
 Apply a two-pole all-pass filter with central frequency (in Hz)
@var{frequency}, and filter-width @var{width}.
 An all-pass filter changes the audio's frequency to phase relationship
 without changing its frequency to amplitude relationship.
 The filter accepts the following options:
@table @option
@item frequency, f
 Set 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.
@end table
@section amerge
 Merge two or more audio streams into a single multi-channel stream.
@ -1071,6 +825,39 @@ amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
 [a2] [b2] amerge
@end example
@section asyncts
 Synchronize audio data with timestamps by squeezing/stretching it and/or
 dropping samples/adding silence when needed.
 This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
 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 atempo
 Adjust audio tempo.
@ -1095,6 +882,236 @@ 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.
@ -1106,6 +1123,149 @@ 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
@end example
@section lowpass
 Apply a low-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 500.
@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 pan
 Mix channels with specific gain levels. The filter accepts the output
@ -1196,6 +1356,11 @@ front left and right:
 pan="stereo: c0=FR : c1=FR"
@end example
@section resample
 Convert the audio sample format, sample rate and channel layout. This filter is
 not meant to be used directly.
@section silencedetect
 Detect silence in an audio stream.
@ -1234,201 +1399,42 @@ ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
@end example
@end itemize
-@section asyncts
+@section treble
 Synchronize audio data with timestamps by squeezing/stretching it and/or
 dropping samples/adding silence when needed.
-This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
+Boost or cut treble (upper) 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:
 The filter accepts the following named parameters:
@table @option
@item gain, g
 Give the gain at whichever is the lower of ~22 kHz and the
 Nyquist frequency. 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 compensate
+@item frequency, f
-Enable stretching/squeezing the data to make it match the timestamps. Disabled
+Set the filter's central frequency and so can be used
-by default. When disabled, time gaps are covered with silence.
+to extend or reduce the frequency range to be boosted or cut.
-
+The default value is @code{3000} Hz.
@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.
@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
-@section atrim
+@item width, w
-Trim the input so that the output contains one continuous subpart of the input.
+Determine how steep is the filter's shelf transition.
 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 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 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 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
@end example
@section resample
 Convert the audio sample format, sample rate and channel layout. This filter is
 not meant to be used directly.
@section volume
 Adjust the input audio volume.