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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
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source
directory, or browsing the online repository at <http://source.ffmpeg.org>.
Maintainers for the specific components are listed in the file MAINTAINERS in the source
code tree.
FFMPEG-RESAMPLER(1)
Generated by $Id: phpMan.php,v 4.55 2007/09/05 04:42:51 chedong Exp $ Author: Che Dong
On Apache
Under GNU General Public License
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