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 format 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,
       option=value for the aresample filter, by setting the value explicitly
       in the "SwrContext" options or using the libavutil/opt.h API for
       programmatic use.

       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
	   in_channel_layout is set.

       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
	   out_channel_layout is set.

       uch, used_channel_count
	   Set the number of used input channels. Default value is 0. 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
	   automatically be chosen when it is not explicitly set.

       icl, in_channel_layout
       ocl, out_channel_layout
	   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.

	   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.

       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 audio 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
	   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'.

       cheby
	   For soxr only, selects passband rolloff none (Chebyshev) & higher-
	   precision 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
	   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.

       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.

       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 between hard (trim/fill) and soft (squeeze/stretch)
	   compensation. Note that all compensation is by default disabled
	   through min_comp.  The default 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-
	   negative 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:

	   none
	       select 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
	   resampling 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
       (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
       the FFmpeg source directory, or browsing the online repository at
       <http://source.ffmpeg.org>.

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

							   FFMPEG-RESAMPLER(1)
