man > FFMPEG-RESAMPLER

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.

       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  automatically  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.

           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
       (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)