@chapter Resampler Options
 @c man begin RESAMPLER OPTIONS
 
 The audio resampler supports the following named options.
 
 Options may be set by specifying -@var{option} @var{value} in the
 FFmpeg tools, @var{option}=@var{value} for the aresample filter,
 by setting the value explicitly in the
 @code{SwrContext} options or using the @file{libavutil/opt.h} API for
 programmatic use.
 
 @table @option
 
 @item ich, in_channel_count
 Set the number of input channels. Default value is 0. Setting this
 value is not mandatory if the corresponding channel layout
 @option{in_channel_layout} is set.
 
 @item och, out_channel_count
 Set the number of output channels. Default value is 0. Setting this
 value is not mandatory if the corresponding channel layout
 @option{out_channel_layout} is set.
 
 @item uch, used_channel_count
 Set the number of used input channels. Default value is 0. This option is
 only used for special remapping.
 
 @item isr, in_sample_rate
 Set the input sample rate. Default value is 0.
 
 @item osr, out_sample_rate
 Set the output sample rate. Default value is 0.
 
 @item isf, in_sample_fmt
 Specify the input sample format. It is set by default to @code{none}.
 
 @item osf, out_sample_fmt
 Specify the output sample format. It is set by default to @code{none}.
 
 @item tsf, internal_sample_fmt
 Set the internal sample format. Default value is @code{none}.
 This will automatically be chosen when it is not explicitly set.
 
 @item icl, in_channel_layout
 @item ocl, out_channel_layout

 Set the input/output channel layout.
 
 See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
 for the required syntax.

 
 @item clev, center_mix_level
 Set the center mix level. It is a value expressed in deciBel, and must be
 in the interval [-32,32].
 
 @item slev, surround_mix_level
 Set the surround mix level. It is a value expressed in deciBel, and must
 be in the interval [-32,32].
 
 @item lfe_mix_level
 Set LFE mix into non LFE level. It is used when there is a LFE input but no
 LFE output. It is a value expressed in deciBel, and must
 be in the interval [-32,32].
 
 @item rmvol, rematrix_volume
 Set rematrix volume. Default value is 1.0.
 

 @item rematrix_maxval
 Set maximum output value for rematrixing.
 This can be used to prevent clipping vs. preventing volumn reduction
 A value of 1.0 prevents cliping.
 

 @item flags, swr_flags
 Set flags used by the converter. Default value is 0.
 
 It supports the following individual flags:
 @table @option
 @item res
 force resampling, this flag forces resampling to be used even when the
 input and output sample rates match.
 @end table
 
 @item dither_scale
 Set the dither scale. Default value is 1.
 
 @item dither_method
 Set dither method. Default value is 0.
 
 Supported values:
 @table @samp
 @item rectangular
 select rectangular dither
 @item triangular
 select triangular dither
 @item triangular_hp
 select triangular dither with high pass
 @item lipshitz
 select lipshitz noise shaping dither
 @item shibata
 select shibata noise shaping dither
 @item low_shibata
 select low shibata noise shaping dither
 @item high_shibata
 select high shibata noise shaping dither
 @item f_weighted
 select f-weighted noise shaping dither
 @item modified_e_weighted
 select modified-e-weighted noise shaping dither
 @item improved_e_weighted
 select improved-e-weighted noise shaping dither
 
 @end table
 
 @item resampler
 Set resampling engine. Default value is swr.
 
 Supported values:
 @table @samp
 @item swr
 select the native SW Resampler; filter options precision and cheby are not
 applicable in this case.
 @item soxr
 select the SoX Resampler (where available); compensation, and filter options
 filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
 case.
 @end table
 
 @item filter_size
 For swr only, set resampling filter size, default value is 32.
 
 @item phase_shift
 For swr only, set resampling phase shift, default value is 10, and must be in
 the interval [0,30].
 
 @item linear_interp
 Use Linear Interpolation if set to 1, default value is 0.
 
 @item cutoff
 Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
 value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
 (which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
 
 @item precision
 For soxr only, the precision in bits to which the resampled signal will be
 calculated.  The default value of 20 (which, with suitable dithering, is
 appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
 value of 28 gives SoX's 'Very High Quality'.
 
 @item cheby
 For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
 approximation for 'irrational' ratios. Default value is 0.
 
 @item async
 For swr only, simple 1 parameter audio sync to timestamps using stretching,
 squeezing, filling and trimming. Setting this to 1 will enable filling and
 trimming, larger values represent the maximum amount in samples that the data
 may be stretched or squeezed for each second.
 Default value is 0, thus no compensation is applied to make the samples match
 the audio timestamps.
 
 @item first_pts
 For swr only, assume the first pts should be this value. The time unit 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 min_comp
 For swr only, set the minimum difference between timestamps and audio data (in
 seconds) to trigger stretching/squeezing/filling or trimming of the
 data to make it match the timestamps. The default is that
 stretching/squeezing/filling and trimming is disabled
 (@option{min_comp} = @code{FLT_MAX}).
 
 @item min_hard_comp
 For swr only, set the minimum difference between timestamps and audio data (in
 seconds) to trigger adding/dropping samples to make it match the
 timestamps.  This option effectively is a threshold to select between
 hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
 all compensation is by default disabled through @option{min_comp}.
 The default is 0.1.
 
 @item comp_duration
 For swr only, set duration (in seconds) over which data is stretched/squeezed
 to make it match the timestamps. Must be a non-negative double float value,
 default value is 1.0.
 
 @item max_soft_comp
 For swr only, set maximum factor by which data is stretched/squeezed to make it
 match the timestamps. Must be a non-negative double float value, default value
 is 0.
 
 @item matrix_encoding
 Select matrixed stereo encoding.
 
 It accepts the following values:
 @table @samp
 @item none
 select none
 @item dolby
 select Dolby
 @item dplii
 select Dolby Pro Logic II
 @end table
 
 Default value is @code{none}.
 
 @item filter_type
 For swr only, select resampling filter type. This only affects resampling
 operations.
 
 It accepts the following values:
 @table @samp
 @item cubic
 select cubic
 @item blackman_nuttall
 select Blackman Nuttall Windowed Sinc
 @item kaiser
 select Kaiser Windowed Sinc
 @end table
 
 @item kaiser_beta
 For swr only, set Kaiser Window Beta value. Must be an integer in the
 interval [2,16], default value is 9.
 

 @item output_sample_bits
 For swr only, set number of used output sample bits for dithering. Must be an integer in the
 interval [0,64], default value is 0, which means it's not used.
 

 @end table
 
 @c man end RESAMPLER OPTIONS