Provided by: ffmpeg_3.3.4-2_amd64 bug

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), ffserver(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)