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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 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
       (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)

ffmpeg 4.4.5 - Generated Sat Nov 23 16:42:24 CST 2024
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