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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 @@ -<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> -<html> -<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ --> - <head> - <meta charset="utf-8"> - <title> - FFmpeg Resampler Documentation - </title> - <meta name="viewport" content="width=device-width,initial-scale=1.0"> - <link rel="stylesheet" type="text/css" href="bootstrap.min.css"> - <link rel="stylesheet" type="text/css" href="style.min.css"> - </head> - <body> - <div class="container"> - <h1> - FFmpeg Resampler Documentation - </h1> -<div align="center"> -</div> - - -<a name="SEC_Top"></a> - -<a name="SEC_Contents"></a> -<h2 class="contents-heading">Table of Contents</h2> - -<div class="contents"> - -<ul class="no-bullet"> - <li><a name="toc-Description" href="#Description">1 Description</a></li> - <li><a name="toc-Resampler-Options" href="#Resampler-Options">2 Resampler Options</a></li> - <li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li> - <li><a name="toc-Authors" href="#Authors">4 Authors</a></li> -</ul> -</div> - - -<a name="Description"></a> -<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2> - -<p>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. -</p> - -<a name="Resampler-Options"></a> -<h2 class="chapter">2 Resampler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Resampler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Resampler-Options" aria-hidden="true">TOC</a></span></h2> - -<p>The audio resampler supports the following named options. -</p> -<p>Options may be set by specifying -<var>option</var> <var>value</var> in the -FFmpeg tools, <var>option</var>=<var>value</var> for the aresample filter, -by setting the value explicitly in the -<code>SwrContext</code> options or using the <samp>libavutil/opt.h</samp> API for -programmatic use. -</p> -<dl compact="compact"> -<dt><samp>ich, in_channel_count</samp></dt> -<dd><p>Set the number of input channels. Default value is 0. Setting this -value is not mandatory if the corresponding channel layout -<samp>in_channel_layout</samp> is set. -</p> -</dd> -<dt><samp>och, out_channel_count</samp></dt> -<dd><p>Set the number of output channels. Default value is 0. Setting this -value is not mandatory if the corresponding channel layout -<samp>out_channel_layout</samp> is set. -</p> -</dd> -<dt><samp>uch, used_channel_count</samp></dt> -<dd><p>Set the number of used input channels. Default value is 0. This option is -only used for special remapping. -</p> -</dd> -<dt><samp>isr, in_sample_rate</samp></dt> -<dd><p>Set the input sample rate. Default value is 0. -</p> -</dd> -<dt><samp>osr, out_sample_rate</samp></dt> -<dd><p>Set the output sample rate. Default value is 0. -</p> -</dd> -<dt><samp>isf, in_sample_fmt</samp></dt> -<dd><p>Specify the input sample format. It is set by default to <code>none</code>. -</p> -</dd> -<dt><samp>osf, out_sample_fmt</samp></dt> -<dd><p>Specify the output sample format. It is set by default to <code>none</code>. -</p> -</dd> -<dt><samp>tsf, internal_sample_fmt</samp></dt> -<dd><p>Set the internal sample format. Default value is <code>none</code>. -This will automatically be chosen when it is not explicitly set. -</p> -</dd> -<dt><samp>icl, in_channel_layout</samp></dt> -<dt><samp>ocl, out_channel_layout</samp></dt> -<dd><p>Set the input/output channel layout. -</p> -<p>See <a href="ffmpeg-utils.html#channel-layout-syntax">(ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual</a> -for the required syntax. -</p> -</dd> -<dt><samp>clev, center_mix_level</samp></dt> -<dd><p>Set the center mix level. It is a value expressed in deciBel, and must be -in the interval [-32,32]. -</p> -</dd> -<dt><samp>slev, surround_mix_level</samp></dt> -<dd><p>Set the surround mix level. It is a value expressed in deciBel, and must -be in the interval [-32,32]. -</p> -</dd> -<dt><samp>lfe_mix_level</samp></dt> -<dd><p>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]. -</p> -</dd> -<dt><samp>rmvol, rematrix_volume</samp></dt> -<dd><p>Set rematrix volume. Default value is 1.0. -</p> -</dd> -<dt><samp>rematrix_maxval</samp></dt> -<dd><p>Set maximum output value for rematrixing. -This can be used to prevent clipping vs. preventing volume reduction. -A value of 1.0 prevents clipping. -</p> -</dd> -<dt><samp>flags, swr_flags</samp></dt> -<dd><p>Set flags used by the converter. Default value is 0. -</p> -<p>It supports the following individual flags: -</p><dl compact="compact"> -<dt><samp>res</samp></dt> -<dd><p>force resampling, this flag forces resampling to be used even when the -input and output sample rates match. -</p></dd> -</dl> - -</dd> -<dt><samp>dither_scale</samp></dt> -<dd><p>Set the dither scale. Default value is 1. -</p> -</dd> -<dt><samp>dither_method</samp></dt> -<dd><p>Set dither method. Default value is 0. -</p> -<p>Supported values: -</p><dl compact="compact"> -<dt>‘<samp>rectangular</samp>’</dt> -<dd><p>select rectangular dither -</p></dd> -<dt>‘<samp>triangular</samp>’</dt> -<dd><p>select triangular dither -</p></dd> -<dt>‘<samp>triangular_hp</samp>’</dt> -<dd><p>select triangular dither with high pass -</p></dd> -<dt>‘<samp>lipshitz</samp>’</dt> -<dd><p>select Lipshitz noise shaping dither. -</p></dd> -<dt>‘<samp>shibata</samp>’</dt> -<dd><p>select Shibata noise shaping dither. -</p></dd> -<dt>‘<samp>low_shibata</samp>’</dt> -<dd><p>select low Shibata noise shaping dither. -</p></dd> -<dt>‘<samp>high_shibata</samp>’</dt> -<dd><p>select high Shibata noise shaping dither. -</p></dd> -<dt>‘<samp>f_weighted</samp>’</dt> -<dd><p>select f-weighted noise shaping dither -</p></dd> -<dt>‘<samp>modified_e_weighted</samp>’</dt> -<dd><p>select modified-e-weighted noise shaping dither -</p></dd> -<dt>‘<samp>improved_e_weighted</samp>’</dt> -<dd><p>select improved-e-weighted noise shaping dither -</p> -</dd> -</dl> - -</dd> -<dt><samp>resampler</samp></dt> -<dd><p>Set resampling engine. Default value is swr. -</p> -<p>Supported values: -</p><dl compact="compact"> -<dt>‘<samp>swr</samp>’</dt> -<dd><p>select the native SW Resampler; filter options precision and cheby are not -applicable in this case. -</p></dd> -<dt>‘<samp>soxr</samp>’</dt> -<dd><p>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. -</p></dd> -</dl> - -</dd> -<dt><samp>filter_size</samp></dt> -<dd><p>For swr only, set resampling filter size, default value is 32. -</p> -</dd> -<dt><samp>phase_shift</samp></dt> -<dd><p>For swr only, set resampling phase shift, default value is 10, and must be in -the interval [0,30]. -</p> -</dd> -<dt><samp>linear_interp</samp></dt> -<dd><p>Use linear interpolation if set to 1, default value is 0. -</p> -</dd> -<dt><samp>exact_rational</samp></dt> -<dd><p>For swr only, when enabled, try to use exact phase_count based on input and -output sample rate. However, if it is larger than <code>1 << phase_shift</code>, -the phase_count will be <code>1 << phase_shift</code> as fallback. Default is disabled. -</p> -</dd> -<dt><samp>cutoff</samp></dt> -<dd><p>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). -</p> -</dd> -<dt><samp>precision</samp></dt> -<dd><p>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’. -</p> -</dd> -<dt><samp>cheby</samp></dt> -<dd><p>For soxr only, selects passband rolloff none (Chebyshev) & higher-precision -approximation for ’irrational’ ratios. Default value is 0. -</p> -</dd> -<dt><samp>async</samp></dt> -<dd><p>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. -</p> -</dd> -<dt><samp>first_pts</samp></dt> -<dd><p>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. -</p> -</dd> -<dt><samp>min_comp</samp></dt> -<dd><p>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 -(<samp>min_comp</samp> = <code>FLT_MAX</code>). -</p> -</dd> -<dt><samp>min_hard_comp</samp></dt> -<dd><p>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 <samp>min_comp</samp>. -The default is 0.1. -</p> -</dd> -<dt><samp>comp_duration</samp></dt> -<dd><p>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. -</p> -</dd> -<dt><samp>max_soft_comp</samp></dt> -<dd><p>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. -</p> -</dd> -<dt><samp>matrix_encoding</samp></dt> -<dd><p>Select matrixed stereo encoding. -</p> -<p>It accepts the following values: -</p><dl compact="compact"> -<dt>‘<samp>none</samp>’</dt> -<dd><p>select none -</p></dd> -<dt>‘<samp>dolby</samp>’</dt> -<dd><p>select Dolby -</p></dd> -<dt>‘<samp>dplii</samp>’</dt> -<dd><p>select Dolby Pro Logic II -</p></dd> -</dl> - -<p>Default value is <code>none</code>. -</p> -</dd> -<dt><samp>filter_type</samp></dt> -<dd><p>For swr only, select resampling filter type. This only affects resampling -operations. -</p> -<p>It accepts the following values: -</p><dl compact="compact"> -<dt>‘<samp>cubic</samp>’</dt> -<dd><p>select cubic -</p></dd> -<dt>‘<samp>blackman_nuttall</samp>’</dt> -<dd><p>select Blackman Nuttall windowed sinc -</p></dd> -<dt>‘<samp>kaiser</samp>’</dt> -<dd><p>select Kaiser windowed sinc -</p></dd> -</dl> - -</dd> -<dt><samp>kaiser_beta</samp></dt> -<dd><p>For swr only, set Kaiser window beta value. Must be a double float value in the -interval [2,16], default value is 9. -</p> -</dd> -<dt><samp>output_sample_bits</samp></dt> -<dd><p>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. -</p> -</dd> -</dl> - - -<a name="See-Also"></a> -<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2> - -<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>, -<a href="libswresample.html">libswresample</a> -</p> - -<a name="Authors"></a> -<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2> - -<p>The FFmpeg developers. -</p> -<p>For details about the authorship, see the Git history of the project -(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command -<code>git log</code> in the FFmpeg source directory, or browsing the -online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>. -</p> -<p>Maintainers for the specific components are listed in the file -<samp>MAINTAINERS</samp> in the source code tree. -</p> - - - <p style="font-size: small;"> - This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>. - </p> - </div> - </body> -</html> |