266 lines
7.7 KiB
Plaintext
266 lines
7.7 KiB
Plaintext
\input texinfo @c -*- texinfo -*-
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@settitle FFmpeg Resampler Documentation
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@titlepage
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@center @titlefont{FFmpeg Resampler Documentation}
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@end titlepage
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@top
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@contents
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@chapter Description
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@c man begin DESCRIPTION
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The FFmpeg resampler provides an high-level interface to the
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libswresample library audio resampling utilities. In particular it
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allows to perform audio resampling, audio channel layout rematrixing,
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and convert audio format and packing layout.
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@c man end DESCRIPTION
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@chapter Resampler Options
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@c man begin RESAMPLER OPTIONS
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The audio resampler supports the following named options.
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Options may be set by specifying -@var{option} @var{value} in the
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FFmpeg tools, @var{option}=@var{value} for the aresample filter,
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by setting the value explicitly in the
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@code{SwrContext} options or using the @file{libavutil/opt.h} API for
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programmatic use.
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@table @option
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@item ich, in_channel_count
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Set the number of input channels. Default value is 0. Setting this
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value is not mandatory if the corresponding channel layout
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@option{in_channel_layout} is set.
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@item och, out_channel_count
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Set the number of output channels. Default value is 0. Setting this
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value is not mandatory if the corresponding channel layout
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@option{out_channel_layout} is set.
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@item uch, used_channel_count
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Set the number of used input channels. Default value is 0. This option is
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only used for special remapping.
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@item isr, in_sample_rate
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Set the input sample rate. Default value is 0.
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@item osr, out_sample_rate
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Set the output sample rate. Default value is 0.
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@item isf, in_sample_fmt
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Specify the input sample format. It is set by default to @code{none}.
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@item osf, out_sample_fmt
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Specify the output sample format. It is set by default to @code{none}.
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@item tsf, internal_sample_fmt
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Set the internal sample format. Default value is @code{none}.
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This will automatically be chosen when it is not explicitly set.
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@item icl, in_channel_layout
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Set the input channel layout.
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@item ocl, out_channel_layout
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Set the output channel layout.
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@item clev, center_mix_level
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Set the center mix level. It is a value expressed in deciBel, and must be
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in the interval [-32,32].
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@item slev, surround_mix_level
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Set the surround mix level. It is a value expressed in deciBel, and must
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be in the interval [-32,32].
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@item lfe_mix_level
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Set LFE mix into non LFE level. It is used when there is a LFE input but no
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LFE output. It is a value expressed in deciBel, and must
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be in the interval [-32,32].
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@item rmvol, rematrix_volume
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Set rematrix volume. Default value is 1.0.
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@item flags, swr_flags
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Set flags used by the converter. Default value is 0.
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It supports the following individual flags:
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@table @option
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@item res
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force resampling, this flag forces resampling to be used even when the
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input and output sample rates match.
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@end table
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@item dither_scale
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Set the dither scale. Default value is 1.
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@item dither_method
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Set dither method. Default value is 0.
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Supported values:
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@table @samp
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@item rectangular
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select rectangular dither
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@item triangular
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select triangular dither
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@item triangular_hp
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select triangular dither with high pass
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@item lipshitz
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select lipshitz noise shaping dither
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@item shibata
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select shibata noise shaping dither
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@item low_shibata
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select low shibata noise shaping dither
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@item high_shibata
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select high shibata noise shaping dither
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@item f_weighted
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select f-weighted noise shaping dither
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@item modified_e_weighted
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select modified-e-weighted noise shaping dither
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@item improved_e_weighted
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select improved-e-weighted noise shaping dither
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@end table
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@item resampler
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Set resampling engine. Default value is swr.
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Supported values:
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@table @samp
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@item swr
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select the native SW Resampler; filter options precision and cheby are not
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applicable in this case.
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@item soxr
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select the SoX Resampler (where available); compensation, and filter options
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filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
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case.
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@end table
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@item filter_size
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For swr only, set resampling filter size, default value is 32.
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@item phase_shift
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For swr only, set resampling phase shift, default value is 10, and must be in
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the interval [0,30].
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@item linear_interp
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Use Linear Interpolation if set to 1, default value is 0.
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@item cutoff
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Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
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value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr
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(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
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@item precision
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For soxr only, the precision in bits to which the resampled signal will be
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calculated. The default value of 20 (which, with suitable dithering, is
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appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
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value of 28 gives SoX's 'Very High Quality'.
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@item cheby
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For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
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approximation for 'irrational' ratios. Default value is 0.
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@item async
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For swr only, simple 1 parameter audio sync to timestamps using stretching,
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squeezing, filling and trimming. Setting this to 1 will enable filling and
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trimming, larger values represent the maximum amount in samples that the data
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may be stretched or squeezed for each second.
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Default value is 0, thus no compensation is applied to make the samples match
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the audio timestamps.
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@item first_pts
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For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
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This allows for padding/trimming at the start of stream. By default, no
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assumption is made about the first frame's expected pts, so no padding or
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trimming is done. For example, this could be set to 0 to pad the beginning with
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silence if an audio stream starts after the video stream or to trim any samples
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with a negative pts due to encoder delay.
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@item min_comp
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For swr only, set the minimum difference between timestamps and audio data (in
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seconds) to trigger stretching/squeezing/filling or trimming of the
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data to make it match the timestamps. The default is that
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stretching/squeezing/filling and trimming is disabled
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(@option{min_comp} = @code{FLT_MAX}).
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@item min_hard_comp
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For swr only, set the minimum difference between timestamps and audio data (in
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seconds) to trigger adding/dropping samples to make it match the
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timestamps. This option effectively is a threshold to select between
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hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
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all compensation is by default disabled through @option{min_comp}.
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The default is 0.1.
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@item comp_duration
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For swr only, set duration (in seconds) over which data is stretched/squeezed
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to make it match the timestamps. Must be a non-negative double float value,
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default value is 1.0.
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@item max_soft_comp
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For swr only, set maximum factor by which data is stretched/squeezed to make it
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match the timestamps. Must be a non-negative double float value, default value
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is 0.
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@item matrix_encoding
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Select matrixed stereo encoding.
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It accepts the following values:
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@table @samp
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@item none
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select none
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@item dolby
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select Dolby
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@item dplii
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select Dolby Pro Logic II
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@end table
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Default value is @code{none}.
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@item filter_type
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For swr only, select resampling filter type. This only affects resampling
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operations.
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It accepts the following values:
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@table @samp
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@item cubic
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select cubic
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@item blackman_nuttall
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select Blackman Nuttall Windowed Sinc
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@item kaiser
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select Kaiser Windowed Sinc
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@end table
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@item kaiser_beta
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For swr only, set Kaiser Window Beta value. Must be an integer in the
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interval [2,16], default value is 9.
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@end table
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@c man end RESAMPLER OPTIONS
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@chapter See Also
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@ifhtml
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@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
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@url{libswresample.html,libswresample}
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@end ifhtml
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@ifnothtml
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ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
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@end ifnothtml
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@include authors.texi
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@ignore
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@setfilename ffmpeg-resampler
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@settitle FFmpeg Resampler
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@end ignore
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@bye
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