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download PDF document: ffmpeg-resampler.1.pdf

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.
uchl, used_chlayout Set used input channel layout. Default is unset. 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.
ichl, in_chlayout ochl, out_chlayout 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]. 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,
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 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