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260 lines
8.4 KiB
Plaintext
260 lines
8.4 KiB
Plaintext
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FFMPEG-RESAMPLER(1) FFMPEG-RESAMPLER(1)
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NAME
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ffmpeg-resampler - FFmpeg Resampler
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DESCRIPTION
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The FFmpeg resampler provides a high-level interface to the
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libswresample library audio resampling utilities. In particular it
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allows one to perform audio resampling, audio channel layout
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rematrixing, and convert audio format and packing layout.
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RESAMPLER OPTIONS
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The audio resampler supports the following named options.
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Options may be set by specifying -option value in the FFmpeg tools,
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option=value for the aresample filter, by setting the value explicitly
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in the "SwrContext" options or using the libavutil/opt.h API for
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programmatic use.
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uchl, used_chlayout
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Set used input channel layout. Default is unset. This option is
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only used for special remapping.
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isr, in_sample_rate
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Set the input sample rate. Default value is 0.
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osr, out_sample_rate
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Set the output sample rate. Default value is 0.
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isf, in_sample_fmt
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Specify the input sample format. It is set by default to "none".
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osf, out_sample_fmt
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Specify the output sample format. It is set by default to "none".
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tsf, internal_sample_fmt
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Set the internal sample format. Default value is "none". This will
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automatically be chosen when it is not explicitly set.
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ichl, in_chlayout
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ochl, out_chlayout
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Set the input/output channel layout.
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See the Channel Layout section in the ffmpeg-utils(1) manual for
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the required syntax.
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clev, center_mix_level
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Set the center mix level. It is a value expressed in deciBel, and
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must be in the interval [-32,32].
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slev, surround_mix_level
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Set the surround mix level. It is a value expressed in deciBel, and
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must be in the interval [-32,32].
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lfe_mix_level
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Set LFE mix into non LFE level. It is used when there is a LFE
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input but no LFE output. It is a value expressed in deciBel, and
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must be in the interval [-32,32].
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rmvol, rematrix_volume
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Set rematrix volume. Default value is 1.0.
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rematrix_maxval
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Set maximum output value for rematrixing. This can be used to
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prevent clipping vs. preventing volume reduction. A value of 1.0
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prevents clipping.
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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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res force resampling, this flag forces resampling to be used even
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when the input and output sample rates match.
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dither_scale
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Set the dither scale. Default value is 1.
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dither_method
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Set dither method. Default value is 0.
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Supported values:
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rectangular
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select rectangular dither
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triangular
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select triangular dither
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triangular_hp
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select triangular dither with high pass
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lipshitz
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select Lipshitz noise shaping dither.
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shibata
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select Shibata noise shaping dither.
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low_shibata
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select low Shibata noise shaping dither.
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high_shibata
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select high Shibata noise shaping dither.
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f_weighted
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select f-weighted noise shaping dither
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modified_e_weighted
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select modified-e-weighted noise shaping dither
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improved_e_weighted
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select improved-e-weighted noise shaping dither
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resampler
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Set resampling engine. Default value is swr.
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Supported values:
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swr select the native SW Resampler; filter options precision and
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cheby are not applicable in this case.
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soxr
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select the SoX Resampler (where available); compensation, and
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filter options filter_size, phase_shift, exact_rational,
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filter_type & kaiser_beta, are not applicable in this case.
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filter_size
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For swr only, set resampling filter size, default value is 32.
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phase_shift
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For swr only, set resampling phase shift, default value is 10, and
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must be in the interval [0,30].
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linear_interp
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Use linear interpolation when enabled (the default). Disable it if
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you want to preserve speed instead of quality when exact_rational
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fails.
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exact_rational
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For swr only, when enabled, try to use exact phase_count based on
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input and output sample rate. However, if it is larger than "1 <<
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phase_shift", the phase_count will be "1 << phase_shift" as
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fallback. Default is enabled.
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cutoff
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Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must
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be a float value between 0 and 1. Default value is 0.97 with swr,
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and 0.91 with soxr (which, with a sample-rate of 44100, preserves
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the entire audio band to 20kHz).
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precision
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For soxr only, the precision in bits to which the resampled signal
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will be calculated. The default value of 20 (which, with suitable
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dithering, is appropriate for a destination bit-depth of 16) gives
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SoX's 'High Quality'; a value of 28 gives SoX's 'Very High
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Quality'.
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cheby
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For soxr only, selects passband rolloff none (Chebyshev) & higher-
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precision approximation for 'irrational' ratios. Default value is
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0.
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async
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For swr only, simple 1 parameter audio sync to timestamps using
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stretching, squeezing, filling and trimming. Setting this to 1 will
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enable filling and trimming, larger values represent the maximum
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amount in samples that the data may be stretched or squeezed for
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each second. Default value is 0, thus no compensation is applied
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to make the samples match the audio timestamps.
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first_pts
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For swr only, assume the first pts should be this value. The time
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unit is 1 / sample rate. This allows for padding/trimming at the
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start of stream. By default, no assumption is made about the first
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frame's expected pts, so no padding or trimming is done. For
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example, this could be set to 0 to pad the beginning with silence
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if an audio stream starts after the video stream or to trim any
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samples with a negative pts due to encoder delay.
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min_comp
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For swr only, set the minimum difference between timestamps and
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audio data (in seconds) to trigger stretching/squeezing/filling or
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trimming of the data to make it match the timestamps. The default
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is that stretching/squeezing/filling and trimming is disabled
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(min_comp = "FLT_MAX").
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min_hard_comp
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For swr only, set the minimum difference between timestamps and
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audio data (in seconds) to trigger adding/dropping samples to make
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it match the timestamps. This option effectively is a threshold to
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select between hard (trim/fill) and soft (squeeze/stretch)
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compensation. Note that all compensation is by default disabled
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through min_comp. The default is 0.1.
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comp_duration
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For swr only, set duration (in seconds) over which data is
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stretched/squeezed to make it match the timestamps. Must be a non-
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negative double float value, default value is 1.0.
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max_soft_comp
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For swr only, set maximum factor by which data is
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stretched/squeezed to make it match the timestamps. Must be a non-
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negative double float value, default value is 0.
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matrix_encoding
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Select matrixed stereo encoding.
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It accepts the following values:
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none
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select none
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dolby
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select Dolby
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dplii
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select Dolby Pro Logic II
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Default value is "none".
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filter_type
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For swr only, select resampling filter type. This only affects
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resampling operations.
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It accepts the following values:
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cubic
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select cubic
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blackman_nuttall
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select Blackman Nuttall windowed sinc
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kaiser
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select Kaiser windowed sinc
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kaiser_beta
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For swr only, set Kaiser window beta value. Must be a double float
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value in the interval [2,16], default value is 9.
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output_sample_bits
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For swr only, set number of used output sample bits for dithering.
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Must be an integer in the interval [0,64], default value is 0,
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which means it's not used.
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SEE ALSO
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ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
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AUTHORS
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The FFmpeg developers.
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For details about the authorship, see the Git history of the project
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(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
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the FFmpeg source directory, or browsing the online repository at
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<https://git.ffmpeg.org/ffmpeg>.
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Maintainers for the specific components are listed in the file
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MAINTAINERS in the source code tree.
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FFMPEG-RESAMPLER(1)
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