353 lines
13 KiB
HTML
353 lines
13 KiB
HTML
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
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<html>
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<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ -->
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<head>
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<meta charset="utf-8">
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<title>
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FFmpeg Resampler Documentation
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</title>
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<meta name="viewport" content="width=device-width,initial-scale=1.0">
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<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
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<link rel="stylesheet" type="text/css" href="style.min.css">
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</head>
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<body>
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<div class="container">
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<h1>
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FFmpeg Resampler Documentation
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</h1>
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<div align="center">
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</div>
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<a name="SEC_Top"></a>
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<div class="Contents_element" id="SEC_Contents">
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<h2 class="contents-heading">Table of Contents</h2>
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<div class="contents">
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<ul class="no-bullet">
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<li><a id="toc-Description" href="#Description">1 Description</a></li>
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<li><a id="toc-Resampler-Options" href="#Resampler-Options">2 Resampler Options</a></li>
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<li><a id="toc-See-Also" href="#See-Also">3 See Also</a></li>
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<li><a id="toc-Authors" href="#Authors">4 Authors</a></li>
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</ul>
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</div>
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</div>
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<a name="Description"></a>
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<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>
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<p>The FFmpeg resampler provides a high-level interface to the
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libswresample library audio resampling utilities. In particular it
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allows one to perform audio resampling, audio channel layout rematrixing,
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and convert audio format and packing layout.
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</p>
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<a name="Resampler-Options"></a>
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<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>
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<p>The audio resampler supports the following named options.
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</p>
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<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
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FFmpeg tools, <var>option</var>=<var>value</var> for the aresample filter,
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by setting the value explicitly in the
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<code>SwrContext</code> options or using the <samp>libavutil/opt.h</samp> API for
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programmatic use.
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</p>
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<dl compact="compact">
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<dt><span><samp>uchl, used_chlayout</samp></span></dt>
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<dd><p>Set used input channel layout. Default is unset. This option is
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only used for special remapping.
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</p>
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</dd>
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<dt><span><samp>isr, in_sample_rate</samp></span></dt>
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<dd><p>Set the input sample rate. Default value is 0.
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</p>
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</dd>
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<dt><span><samp>osr, out_sample_rate</samp></span></dt>
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<dd><p>Set the output sample rate. Default value is 0.
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</p>
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</dd>
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<dt><span><samp>isf, in_sample_fmt</samp></span></dt>
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<dd><p>Specify the input sample format. It is set by default to <code>none</code>.
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</p>
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</dd>
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<dt><span><samp>osf, out_sample_fmt</samp></span></dt>
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<dd><p>Specify the output sample format. It is set by default to <code>none</code>.
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</p>
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</dd>
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<dt><span><samp>tsf, internal_sample_fmt</samp></span></dt>
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<dd><p>Set the internal sample format. Default value is <code>none</code>.
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This will automatically be chosen when it is not explicitly set.
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</p>
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</dd>
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<dt><span><samp>ichl, in_chlayout</samp></span></dt>
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<dt><span><samp>ochl, out_chlayout</samp></span></dt>
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<dd><p>Set the input/output channel layout.
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</p>
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<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>
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for the required syntax.
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</p>
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</dd>
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<dt><span><samp>clev, center_mix_level</samp></span></dt>
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<dd><p>Set the center mix level. It is a value expressed in deciBel, and must be
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in the interval [-32,32].
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</p>
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</dd>
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<dt><span><samp>slev, surround_mix_level</samp></span></dt>
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<dd><p>Set the surround mix level. It is a value expressed in deciBel, and must
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be in the interval [-32,32].
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</p>
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</dd>
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<dt><span><samp>lfe_mix_level</samp></span></dt>
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<dd><p>Set LFE mix into non LFE level. It is used when there is a LFE input but no
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LFE output. It is a value expressed in deciBel, and must
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be in the interval [-32,32].
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</p>
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</dd>
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<dt><span><samp>rmvol, rematrix_volume</samp></span></dt>
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<dd><p>Set rematrix volume. Default value is 1.0.
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</p>
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</dd>
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<dt><span><samp>rematrix_maxval</samp></span></dt>
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<dd><p>Set maximum output value for rematrixing.
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This can be used to prevent clipping vs. preventing volume reduction.
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A value of 1.0 prevents clipping.
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</p>
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</dd>
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<dt><span><samp>flags, swr_flags</samp></span></dt>
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<dd><p>Set flags used by the converter. Default value is 0.
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</p>
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<p>It supports the following individual flags:
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</p><dl compact="compact">
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<dt><span><samp>res</samp></span></dt>
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<dd><p>force resampling, this flag forces resampling to be used even when the
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input and output sample rates match.
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</p></dd>
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</dl>
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</dd>
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<dt><span><samp>dither_scale</samp></span></dt>
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<dd><p>Set the dither scale. Default value is 1.
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</p>
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</dd>
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<dt><span><samp>dither_method</samp></span></dt>
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<dd><p>Set dither method. Default value is 0.
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</p>
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<p>Supported values:
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</p><dl compact="compact">
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<dt><span>‘<samp>rectangular</samp>’</span></dt>
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<dd><p>select rectangular dither
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</p></dd>
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<dt><span>‘<samp>triangular</samp>’</span></dt>
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<dd><p>select triangular dither
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</p></dd>
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<dt><span>‘<samp>triangular_hp</samp>’</span></dt>
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<dd><p>select triangular dither with high pass
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</p></dd>
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<dt><span>‘<samp>lipshitz</samp>’</span></dt>
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<dd><p>select Lipshitz noise shaping dither.
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</p></dd>
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<dt><span>‘<samp>shibata</samp>’</span></dt>
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<dd><p>select Shibata noise shaping dither.
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</p></dd>
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<dt><span>‘<samp>low_shibata</samp>’</span></dt>
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<dd><p>select low Shibata noise shaping dither.
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</p></dd>
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<dt><span>‘<samp>high_shibata</samp>’</span></dt>
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<dd><p>select high Shibata noise shaping dither.
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</p></dd>
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<dt><span>‘<samp>f_weighted</samp>’</span></dt>
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<dd><p>select f-weighted noise shaping dither
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</p></dd>
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<dt><span>‘<samp>modified_e_weighted</samp>’</span></dt>
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<dd><p>select modified-e-weighted noise shaping dither
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</p></dd>
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<dt><span>‘<samp>improved_e_weighted</samp>’</span></dt>
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<dd><p>select improved-e-weighted noise shaping dither
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</p>
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</dd>
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</dl>
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</dd>
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<dt><span><samp>resampler</samp></span></dt>
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<dd><p>Set resampling engine. Default value is swr.
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</p>
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<p>Supported values:
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</p><dl compact="compact">
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<dt><span>‘<samp>swr</samp>’</span></dt>
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<dd><p>select the native SW Resampler; filter options precision and cheby are not
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applicable in this case.
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</p></dd>
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<dt><span>‘<samp>soxr</samp>’</span></dt>
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<dd><p>select the SoX Resampler (where available); compensation, and filter options
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filter_size, phase_shift, exact_rational, filter_type & kaiser_beta, are not
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applicable in this case.
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</p></dd>
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</dl>
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</dd>
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<dt><span><samp>filter_size</samp></span></dt>
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<dd><p>For swr only, set resampling filter size, default value is 32.
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</p>
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</dd>
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<dt><span><samp>phase_shift</samp></span></dt>
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<dd><p>For swr only, set resampling phase shift, default value is 10, and must be in
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the interval [0,30].
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</p>
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</dd>
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<dt><span><samp>linear_interp</samp></span></dt>
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<dd><p>Use linear interpolation when enabled (the default). Disable it if you want
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to preserve speed instead of quality when exact_rational fails.
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</p>
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</dd>
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<dt><span><samp>exact_rational</samp></span></dt>
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<dd><p>For swr only, when enabled, try to use exact phase_count based on input and
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output sample rate. However, if it is larger than <code>1 << phase_shift</code>,
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the phase_count will be <code>1 << phase_shift</code> as fallback. Default is enabled.
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</p>
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</dd>
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<dt><span><samp>cutoff</samp></span></dt>
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<dd><p>Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
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value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr
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(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
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</p>
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</dd>
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<dt><span><samp>precision</samp></span></dt>
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<dd><p>For soxr only, the precision in bits to which the resampled signal will be
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calculated. The default value of 20 (which, with suitable dithering, is
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appropriate for a destination bit-depth of 16) gives SoX’s ’High Quality’; a
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value of 28 gives SoX’s ’Very High Quality’.
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</p>
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</dd>
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<dt><span><samp>cheby</samp></span></dt>
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<dd><p>For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
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approximation for ’irrational’ ratios. Default value is 0.
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</p>
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</dd>
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<dt><span><samp>async</samp></span></dt>
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<dd><p>For swr only, simple 1 parameter audio sync to timestamps using stretching,
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squeezing, filling and trimming. Setting this to 1 will enable filling and
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trimming, larger values represent the maximum amount in samples that the data
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may be stretched or squeezed for each second.
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Default value is 0, thus no compensation is applied to make the samples match
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the audio timestamps.
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</p>
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</dd>
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<dt><span><samp>first_pts</samp></span></dt>
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<dd><p>For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
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This allows for padding/trimming at the start of stream. By default, no
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assumption is made about the first frame’s expected pts, so no padding or
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trimming is done. For example, this could be set to 0 to pad the beginning with
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silence if an audio stream starts after the video stream or to trim any samples
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with a negative pts due to encoder delay.
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</p>
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</dd>
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<dt><span><samp>min_comp</samp></span></dt>
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<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
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seconds) to trigger stretching/squeezing/filling or trimming of the
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data to make it match the timestamps. The default is that
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stretching/squeezing/filling and trimming is disabled
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(<samp>min_comp</samp> = <code>FLT_MAX</code>).
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</p>
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</dd>
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<dt><span><samp>min_hard_comp</samp></span></dt>
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<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
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seconds) to trigger adding/dropping samples to make it match the
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timestamps. This option effectively is a threshold to select between
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hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
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all compensation is by default disabled through <samp>min_comp</samp>.
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The default is 0.1.
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</p>
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</dd>
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<dt><span><samp>comp_duration</samp></span></dt>
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<dd><p>For swr only, set duration (in seconds) over which data is stretched/squeezed
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to make it match the timestamps. Must be a non-negative double float value,
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default value is 1.0.
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</p>
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</dd>
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<dt><span><samp>max_soft_comp</samp></span></dt>
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<dd><p>For swr only, set maximum factor by which data is stretched/squeezed to make it
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match the timestamps. Must be a non-negative double float value, default value
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is 0.
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</p>
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</dd>
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<dt><span><samp>matrix_encoding</samp></span></dt>
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<dd><p>Select matrixed stereo encoding.
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</p>
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<p>It accepts the following values:
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</p><dl compact="compact">
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<dt><span>‘<samp>none</samp>’</span></dt>
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<dd><p>select none
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</p></dd>
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<dt><span>‘<samp>dolby</samp>’</span></dt>
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<dd><p>select Dolby
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</p></dd>
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<dt><span>‘<samp>dplii</samp>’</span></dt>
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<dd><p>select Dolby Pro Logic II
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</p></dd>
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</dl>
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<p>Default value is <code>none</code>.
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</p>
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</dd>
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<dt><span><samp>filter_type</samp></span></dt>
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<dd><p>For swr only, select resampling filter type. This only affects resampling
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operations.
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</p>
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<p>It accepts the following values:
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</p><dl compact="compact">
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<dt><span>‘<samp>cubic</samp>’</span></dt>
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<dd><p>select cubic
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</p></dd>
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<dt><span>‘<samp>blackman_nuttall</samp>’</span></dt>
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<dd><p>select Blackman Nuttall windowed sinc
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</p></dd>
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<dt><span>‘<samp>kaiser</samp>’</span></dt>
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<dd><p>select Kaiser windowed sinc
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</p></dd>
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</dl>
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</dd>
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<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
|
||
|
|
(https://git.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="https://git.ffmpeg.org/ffmpeg">https://git.ffmpeg.org/ffmpeg</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>
|