# phpman > man > ffmpeg-resampler(1)

[FFMPEG-RESAMPLER(1)](https://www.chedong.com/phpMan.php/man/FFMPEG-RESAMPLER/1/markdown)                                                              [FFMPEG-RESAMPLER(1)](https://www.chedong.com/phpMan.php/man/FFMPEG-RESAMPLER/1/markdown)



## 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)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown)** **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)](https://www.chedong.com/phpMan.php/man/ffmpeg/1/markdown), [**ffplay**(1)](https://www.chedong.com/phpMan.php/man/ffplay/1/markdown), [**ffprobe**(1)](https://www.chedong.com/phpMan.php/man/ffprobe/1/markdown), [**libswresample**(3)](https://www.chedong.com/phpMan.php/man/libswresample/3/markdown)

## 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)](https://www.chedong.com/phpMan.php/man/FFMPEG-RESAMPLER/1/markdown)