Description
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The FFmpeg resampler provides an high-level interface to the libswresample library audio resampling utilities. In particular it allows to perform audio resampling, audio channel layout rematrixing, and convert audio format and packing layout.

Resampler Options
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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 choosen when its not explicitly set.

icl, in_channel_layout
     Set the input channel layout.

ocl, out_channel_layout
     Set the output channel layout.

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.

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, 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 if set to 1, default value is 0.

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 an integer in the interval [2,16], default value is 9.

See Also
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ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)

     [file authors.texi not included]