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diff --git a/ffmpeg/doc/ffmpeg-resampler.html b/ffmpeg/doc/ffmpeg-resampler.html new file mode 100644 index 0000000..8a11b08 --- /dev/null +++ b/ffmpeg/doc/ffmpeg-resampler.html @@ -0,0 +1,364 @@ +<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> +<html> +<!-- Created by GNU Texinfo 6.8, https://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> + +<div class="Contents_element" id="SEC_Contents"> +<h2 class="contents-heading">Table of Contents</h2> + +<div class="contents"> + +<ul class="no-bullet"> + <li><a id="toc-Description" href="#Description">1 Description</a></li> + <li><a id="toc-Resampler-Options" href="#Resampler-Options">2 Resampler Options</a></li> + <li><a id="toc-See-Also" href="#See-Also">3 See Also</a></li> + <li><a id="toc-Authors" href="#Authors">4 Authors</a></li> +</ul> +</div> +</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><span><samp>ich, in_channel_count</samp></span></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><span><samp>och, out_channel_count</samp></span></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><span><samp>uch, used_channel_count</samp></span></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><span><samp>isr, in_sample_rate</samp></span></dt> +<dd><p>Set the input sample rate. Default value is 0. +</p> +</dd> +<dt><span><samp>osr, out_sample_rate</samp></span></dt> +<dd><p>Set the output sample rate. Default value is 0. +</p> +</dd> +<dt><span><samp>isf, in_sample_fmt</samp></span></dt> +<dd><p>Specify the input sample format. It is set by default to <code>none</code>. +</p> +</dd> +<dt><span><samp>osf, out_sample_fmt</samp></span></dt> +<dd><p>Specify the output sample format. It is set by default to <code>none</code>. +</p> +</dd> +<dt><span><samp>tsf, internal_sample_fmt</samp></span></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><span><samp>icl, in_channel_layout</samp></span></dt> +<dt><span><samp>ocl, out_channel_layout</samp></span></dt> +<dd><p>Set the input/output channel layout. +</p> +<p>See <a data-manual="ffmpeg-utils" 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><span><samp>clev, center_mix_level</samp></span></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><span><samp>slev, surround_mix_level</samp></span></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><span><samp>lfe_mix_level</samp></span></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><span><samp>rmvol, rematrix_volume</samp></span></dt> +<dd><p>Set rematrix volume. Default value is 1.0. +</p> +</dd> +<dt><span><samp>rematrix_maxval</samp></span></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><span><samp>flags, swr_flags</samp></span></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><span><samp>res</samp></span></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><span><samp>dither_scale</samp></span></dt> +<dd><p>Set the dither scale. Default value is 1. +</p> +</dd> +<dt><span><samp>dither_method</samp></span></dt> +<dd><p>Set dither method. Default value is 0. +</p> +<p>Supported values: +</p><dl compact="compact"> +<dt><span>‘<samp>rectangular</samp>’</span></dt> +<dd><p>select rectangular dither +</p></dd> +<dt><span>‘<samp>triangular</samp>’</span></dt> +<dd><p>select triangular dither +</p></dd> +<dt><span>‘<samp>triangular_hp</samp>’</span></dt> +<dd><p>select triangular dither with high pass +</p></dd> +<dt><span>‘<samp>lipshitz</samp>’</span></dt> +<dd><p>select Lipshitz noise shaping dither. +</p></dd> +<dt><span>‘<samp>shibata</samp>’</span></dt> +<dd><p>select Shibata noise shaping dither. +</p></dd> +<dt><span>‘<samp>low_shibata</samp>’</span></dt> +<dd><p>select low Shibata noise shaping dither. +</p></dd> +<dt><span>‘<samp>high_shibata</samp>’</span></dt> +<dd><p>select high Shibata noise shaping dither. +</p></dd> +<dt><span>‘<samp>f_weighted</samp>’</span></dt> +<dd><p>select f-weighted noise shaping dither +</p></dd> +<dt><span>‘<samp>modified_e_weighted</samp>’</span></dt> +<dd><p>select modified-e-weighted noise shaping dither +</p></dd> +<dt><span>‘<samp>improved_e_weighted</samp>’</span></dt> +<dd><p>select improved-e-weighted noise shaping dither +</p> +</dd> +</dl> + +</dd> +<dt><span><samp>resampler</samp></span></dt> +<dd><p>Set resampling engine. Default value is swr. +</p> +<p>Supported values: +</p><dl compact="compact"> +<dt><span>‘<samp>swr</samp>’</span></dt> +<dd><p>select the native SW Resampler; filter options precision and cheby are not +applicable in this case. +</p></dd> +<dt><span>‘<samp>soxr</samp>’</span></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><span><samp>filter_size</samp></span></dt> +<dd><p>For swr only, set resampling filter size, default value is 32. +</p> +</dd> +<dt><span><samp>phase_shift</samp></span></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><span><samp>linear_interp</samp></span></dt> +<dd><p>Use linear interpolation when enabled (the default). Disable it if you want +to preserve speed instead of quality when exact_rational fails. +</p> +</dd> +<dt><span><samp>exact_rational</samp></span></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 enabled. +</p> +</dd> +<dt><span><samp>cutoff</samp></span></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><span><samp>precision</samp></span></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><span><samp>cheby</samp></span></dt> +<dd><p>For soxr only, selects passband rolloff none (Chebyshev) & higher-precision +approximation for ’irrational’ ratios. Default value is 0. +</p> +</dd> +<dt><span><samp>async</samp></span></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><span><samp>first_pts</samp></span></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><span><samp>min_comp</samp></span></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><span><samp>min_hard_comp</samp></span></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><span><samp>comp_duration</samp></span></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><span><samp>max_soft_comp</samp></span></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><span><samp>matrix_encoding</samp></span></dt> +<dd><p>Select matrixed stereo encoding. +</p> +<p>It accepts the following values: +</p><dl compact="compact"> +<dt><span>‘<samp>none</samp>’</span></dt> +<dd><p>select none +</p></dd> +<dt><span>‘<samp>dolby</samp>’</span></dt> +<dd><p>select Dolby +</p></dd> +<dt><span>‘<samp>dplii</samp>’</span></dt> +<dd><p>select Dolby Pro Logic II +</p></dd> +</dl> + +<p>Default value is <code>none</code>. +</p> +</dd> +<dt><span><samp>filter_type</samp></span></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><span>‘<samp>cubic</samp>’</span></dt> +<dd><p>select cubic +</p></dd> +<dt><span>‘<samp>blackman_nuttall</samp>’</span></dt> +<dd><p>select Blackman Nuttall windowed sinc +</p></dd> +<dt><span>‘<samp>kaiser</samp>’</span></dt> +<dd><p>select Kaiser windowed sinc +</p></dd> +</dl> + +</dd> +<dt><span><samp>kaiser_beta</samp></span></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><span><samp>output_sample_bits</samp></span></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="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="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>. + </p> + </div> + </body> +</html> |