From 734eee1af2c21976e8f57c4ca498593a305fb22e Mon Sep 17 00:00:00 2001 From: Xavier Del Campo Romero Date: Sun, 3 Jan 2021 02:06:58 +0100 Subject: Remove ffmpeg binary from project --- Music/ffmpeg/doc/ffmpeg-resampler.html | 364 --------------------------------- 1 file changed, 364 deletions(-) delete mode 100644 Music/ffmpeg/doc/ffmpeg-resampler.html (limited to 'Music/ffmpeg/doc/ffmpeg-resampler.html') diff --git a/Music/ffmpeg/doc/ffmpeg-resampler.html b/Music/ffmpeg/doc/ffmpeg-resampler.html deleted file mode 100644 index a362353..0000000 --- a/Music/ffmpeg/doc/ffmpeg-resampler.html +++ /dev/null @@ -1,364 +0,0 @@ - - - - - - - FFmpeg Resampler Documentation - - - - - - -
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- FFmpeg Resampler Documentation -

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-
- - - - - -

Table of Contents

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- - -
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1 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. -

- - -

2 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. -

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uch, used_channel_count
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Set the number of used input channels. Default value is 0. This option is -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 automatically be chosen when it is not explicitly set. -

-
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icl, in_channel_layout
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ocl, out_channel_layout
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Set the input/output channel layout. -

-

See (ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual -for the required syntax. -

-
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clev, center_mix_level
-

Set the center mix level. It is a value expressed in deciBel, and must be -in the interval [-32,32]. -

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slev, surround_mix_level
-

Set the surround mix level. It is a value expressed in deciBel, and must -be in the interval [-32,32]. -

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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]. -

-
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rmvol, rematrix_volume
-

Set rematrix volume. Default value is 1.0. -

-
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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. -

-
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flags, swr_flags
-

Set flags used by the converter. Default value is 0. -

-

It supports the following individual flags: -

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res
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force resampling, this flag forces resampling to be used even 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’
-

select rectangular dither -

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‘triangular’
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select triangular dither -

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‘triangular_hp’
-

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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-
- -
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resampler
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Set resampling engine. Default value is swr. -

-

Supported values: -

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‘swr’
-

select the native SW Resampler; filter options precision and cheby are not -applicable in this case. -

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‘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. -

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- -
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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
-

For swr only, set resampling phase shift, default value is 10, and must be in -the interval [0,30]. -

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linear_interp
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Use linear interpolation if set to 1, default value is 0. -

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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 disabled. -

-
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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). -

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precision
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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’. -

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cheby
-

For soxr only, selects passband rolloff none (Chebyshev) & higher-precision -approximation for ’irrational’ ratios. Default value is 0. -

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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. -

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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. -

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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). -

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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. -

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comp_duration
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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. -

-
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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. -

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matrix_encoding
-

Select matrixed stereo encoding. -

-

It accepts the following values: -

-
‘none’
-

select none -

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‘dolby’
-

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
-

For swr only, select resampling filter type. This only affects resampling -operations. -

-

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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- -
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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. -

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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. -

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3 See Also

- -

ffmpeg, ffplay, ffprobe, ffserver, -libswresample -

- - -

4 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. -

- - -

- This document was generated using makeinfo. -

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- - -- cgit v1.2.3