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+@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 volume reduction.
+A value of 1.0 prevents clipping.
+
+@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, exact_rational, 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 exact_rational
+For swr only, when enabled, try to use exact phase_count based on input and
+output sample rate. However, if it is larger than @code{1 << phase_shift},
+the phase_count will be @code{1 << phase_shift} as fallback. Default is disabled.
+
+@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 a double float value 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