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FFMPEG-RESAMPLER(1)					     FFMPEG-RESAMPLER(1)

NAME
     ffmpeg-resampler - FFmpeg Resampler

DESCRIPTION
     The FFmpeg resampler provides a high-level interface to the libswresample
     library audio resampling utilities. In particular it allows one to perform
     audio resampling, audio channel layout rematrixing, and convert audio for-
     mat and packing layout.

RESAMPLER OPTIONS
     The audio resampler supports the following named options.

     Options may be set by specifying -option value in the FFmpeg tools, op-
     tion=value for the aresample filter, by setting the value explicitly in the
     "SwrContext" options or using the libavutil/opt.h API for programmatic use.

     uchl, used_chlayout
	 Set used input channel layout. Default is unset. This option is only
	 used for special remapping.

     isr, in_sample_rate
	 Set the input sample rate. Default value is 0.

     osr, out_sample_rate
	 Set the output sample rate. Default value is 0.

     isf, in_sample_fmt
	 Specify the input sample format. It is set by default to "none".

     osf, out_sample_fmt
	 Specify the output sample format. It is set by default to "none".

     tsf, internal_sample_fmt
	 Set the internal sample format. Default value is "none".  This will au-
	 tomatically be chosen when it is not explicitly set.

     ichl, in_chlayout
     ochl, out_chlayout
	 Set the input/output channel layout.

	 See the Channel Layout section in the ffmpeg-utils(1) manual for the
	 required syntax.

     clev, center_mix_level
	 Set the center mix level. It is a value expressed in deciBel, and must
	 be in the interval [-32,32].

     slev, surround_mix_level
	 Set the surround mix level. It is a value expressed in deciBel, and
	 must be in the interval [-32,32].

     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].

     rmvol, rematrix_volume
	 Set rematrix volume. Default value is 1.0.

     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.

     flags, swr_flags
	 Set flags used by the converter. Default value is 0.

	 It supports the following individual flags:

	 res force resampling, this flag forces resampling to be used even when
	     the input and output sample rates match.

     dither_scale
	 Set the dither scale. Default value is 1.

     dither_method
	 Set dither method. Default value is 0.

	 Supported values:

	 rectangular
	     select rectangular dither

	 triangular
	     select triangular dither

	 triangular_hp
	     select triangular dither with high pass

	 lipshitz
	     select Lipshitz noise shaping dither.

	 shibata
	     select Shibata noise shaping dither.

	 low_shibata
	     select low Shibata noise shaping dither.

	 high_shibata
	     select high Shibata noise shaping dither.

	 f_weighted
	     select f-weighted noise shaping dither

	 modified_e_weighted
	     select modified-e-weighted noise shaping dither

	 improved_e_weighted
	     select improved-e-weighted noise shaping dither

     resampler
	 Set resampling engine. Default value is swr.

	 Supported values:

	 swr select the native SW Resampler; filter options precision and cheby
	     are not applicable in this case.

	 soxrselect the SoX Resampler (where available); compensation, and fil-
	     ter options filter_size, phase_shift, exact_rational, filter_type &
	     kaiser_beta, are not applicable in this case.

     filter_size
	 For swr only, set resampling filter size, default value is 32.

     phase_shift
	 For swr only, set resampling phase shift, default value is 10, and must
	 be in the interval [0,30].

     linear_interp
	 Use linear interpolation when enabled (the default). Disable it if you
	 want to preserve speed instead of quality when exact_rational fails.

     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 "1 <<
	 phase_shift", the phase_count will be "1 << phase_shift" as fallback.
	 Default is enabled.

     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 au-
	 dio band to 20kHz).

     precision
	 For soxr only, the precision in bits to which the resampled signal will
	 be calculated.  The default value of 20 (which, with suitable dither-
	 ing, is appropriate for a destination bit-depth of 16) gives SoX's
	 'High Quality'; a value of 28 gives SoX's 'Very High Quality'.

     cheby
	 For soxr only, selects passband rolloff none (Chebyshev) & higher-pre-
	 cision approximation for 'irrational' ratios. Default value is 0.

     async
	 For swr only, simple 1 parameter audio sync to timestamps using
	 stretching, squeezing, filling and trimming. Setting this to 1 will en-
	 able 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.

     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 ex-
	 pected 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.

     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 (min_comp =
	 "FLT_MAX").

     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 be-
	 tween hard (trim/fill) and soft (squeeze/stretch) compensation. Note
	 that all compensation is by default disabled through min_comp.  The de-
	 fault is 0.1.

     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-nega-
	 tive double float value, default value is 1.0.

     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.

     matrix_encoding
	 Select matrixed stereo encoding.

	 It accepts the following values:

	 noneselect none

	 dolby
	     select Dolby

	 dplii
	     select Dolby Pro Logic II

	 Default value is "none".

     filter_type
	 For swr only, select resampling filter type. This only affects resam-
	 pling operations.

	 It accepts the following values:

	 cubic
	     select cubic

	 blackman_nuttall
	     select Blackman Nuttall windowed sinc

	 kaiser
	     select Kaiser windowed sinc

     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.

     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.

SEE ALSO
     ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)

AUTHORS
     The FFmpeg developers.

     For details about the authorship, see the Git history of the project
     (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the
     FFmpeg source directory, or browsing the online repository at
     <https://git.ffmpeg.org/ffmpeg>.

     Maintainers for the specific components are listed in the file MAINTAINERS
     in the source code tree.

							     FFMPEG-RESAMPLER(1)

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