2569 lines
101 KiB
HTML
2569 lines
101 KiB
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<title>
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FFmpeg Protocols 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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<h1>
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FFmpeg Protocols 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-Protocol-Options" href="#Protocol-Options">2 Protocol Options</a></li>
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<li><a id="toc-Protocols" href="#Protocols">3 Protocols</a>
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<ul class="no-bullet">
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<li><a id="toc-amqp" href="#amqp">3.1 amqp</a></li>
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<li><a id="toc-async" href="#async">3.2 async</a></li>
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<li><a id="toc-bluray" href="#bluray">3.3 bluray</a></li>
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<li><a id="toc-cache" href="#cache">3.4 cache</a></li>
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<li><a id="toc-concat" href="#concat">3.5 concat</a></li>
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<li><a id="toc-concatf" href="#concatf">3.6 concatf</a></li>
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<li><a id="toc-crypto" href="#crypto">3.7 crypto</a></li>
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<li><a id="toc-data" href="#data">3.8 data</a></li>
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<li><a id="toc-fd" href="#fd">3.9 fd</a></li>
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<li><a id="toc-file" href="#file">3.10 file</a></li>
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<li><a id="toc-ftp" href="#ftp">3.11 ftp</a></li>
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<li><a id="toc-gopher" href="#gopher">3.12 gopher</a></li>
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<li><a id="toc-gophers" href="#gophers">3.13 gophers</a></li>
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<li><a id="toc-hls" href="#hls">3.14 hls</a></li>
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<li><a id="toc-http" href="#http">3.15 http</a>
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<ul class="no-bullet">
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<li><a id="toc-HTTP-Cookies" href="#HTTP-Cookies">3.15.1 HTTP Cookies</a></li>
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</ul></li>
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<li><a id="toc-Icecast" href="#Icecast">3.16 Icecast</a></li>
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<li><a id="toc-ipfs" href="#ipfs">3.17 ipfs</a></li>
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<li><a id="toc-mmst" href="#mmst">3.18 mmst</a></li>
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<li><a id="toc-mmsh" href="#mmsh">3.19 mmsh</a></li>
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<li><a id="toc-md5" href="#md5">3.20 md5</a></li>
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<li><a id="toc-pipe" href="#pipe">3.21 pipe</a></li>
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<li><a id="toc-prompeg" href="#prompeg">3.22 prompeg</a></li>
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<li><a id="toc-rist" href="#rist">3.23 rist</a></li>
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<li><a id="toc-rtmp" href="#rtmp">3.24 rtmp</a></li>
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<li><a id="toc-rtmpe" href="#rtmpe">3.25 rtmpe</a></li>
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<li><a id="toc-rtmps" href="#rtmps">3.26 rtmps</a></li>
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<li><a id="toc-rtmpt" href="#rtmpt">3.27 rtmpt</a></li>
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<li><a id="toc-rtmpte" href="#rtmpte">3.28 rtmpte</a></li>
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<li><a id="toc-rtmpts" href="#rtmpts">3.29 rtmpts</a></li>
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<li><a id="toc-libsmbclient" href="#libsmbclient">3.30 libsmbclient</a></li>
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<li><a id="toc-libssh" href="#libssh">3.31 libssh</a></li>
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<li><a id="toc-librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte" href="#librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte">3.32 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte</a></li>
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<li><a id="toc-rtp" href="#rtp">3.33 rtp</a></li>
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<li><a id="toc-rtsp" href="#rtsp">3.34 rtsp</a>
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<ul class="no-bullet">
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<li><a id="toc-Muxer" href="#Muxer">3.34.1 Muxer</a></li>
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<li><a id="toc-Demuxer" href="#Demuxer">3.34.2 Demuxer</a></li>
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<li><a id="toc-Examples" href="#Examples">3.34.3 Examples</a></li>
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</ul></li>
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<li><a id="toc-sap" href="#sap">3.35 sap</a>
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<ul class="no-bullet">
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<li><a id="toc-Muxer-1" href="#Muxer-1">3.35.1 Muxer</a></li>
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<li><a id="toc-Demuxer-1" href="#Demuxer-1">3.35.2 Demuxer</a></li>
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</ul></li>
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<li><a id="toc-sctp" href="#sctp">3.36 sctp</a></li>
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<li><a id="toc-srt" href="#srt">3.37 srt</a></li>
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<li><a id="toc-srtp" href="#srtp">3.38 srtp</a></li>
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<li><a id="toc-subfile" href="#subfile">3.39 subfile</a></li>
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<li><a id="toc-tee" href="#tee">3.40 tee</a></li>
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<li><a id="toc-tcp" href="#tcp">3.41 tcp</a></li>
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<li><a id="toc-tls" href="#tls">3.42 tls</a></li>
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<li><a id="toc-udp" href="#udp">3.43 udp</a>
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<ul class="no-bullet">
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<li><a id="toc-Examples-1" href="#Examples-1">3.43.1 Examples</a></li>
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</ul></li>
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<li><a id="toc-unix" href="#unix">3.44 unix</a></li>
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<li><a id="toc-zmq" href="#zmq">3.45 zmq</a></li>
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</ul></li>
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<li><a id="toc-See-Also" href="#See-Also">4 See Also</a></li>
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<li><a id="toc-Authors" href="#Authors">5 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>This document describes the input and output protocols provided by the
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libavformat library.
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</p>
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<a name="Protocol-Options"></a>
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<h2 class="chapter">2 Protocol Options<span class="pull-right"><a class="anchor hidden-xs" href="#Protocol-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Protocol-Options" aria-hidden="true">TOC</a></span></h2>
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<p>The libavformat library provides some generic global options, which
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can be set on all the protocols. In addition each protocol may support
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so-called private options, which are specific for that component.
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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, or by setting the value explicitly in the
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<code>AVFormatContext</code> options or using the <samp>libavutil/opt.h</samp> API
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for programmatic use.
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</p>
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<p>The list of supported options follows:
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</p>
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<dl compact="compact">
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<dt><span><samp>protocol_whitelist <var>list</var> (<em>input</em>)</samp></span></dt>
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<dd><p>Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
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prefixed by "-" are disabled.
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All protocols are allowed by default but protocols used by an another
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protocol (nested protocols) are restricted to a per protocol subset.
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</p></dd>
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</dl>
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<a name="Protocols"></a>
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<h2 class="chapter">3 Protocols<span class="pull-right"><a class="anchor hidden-xs" href="#Protocols" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Protocols" aria-hidden="true">TOC</a></span></h2>
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<p>Protocols are configured elements in FFmpeg that enable access to
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resources that require specific protocols.
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</p>
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<p>When you configure your FFmpeg build, all the supported protocols are
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enabled by default. You can list all available ones using the
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configure option "–list-protocols".
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</p>
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<p>You can disable all the protocols using the configure option
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"–disable-protocols", and selectively enable a protocol using the
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option "–enable-protocol=<var>PROTOCOL</var>", or you can disable a
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particular protocol using the option
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"–disable-protocol=<var>PROTOCOL</var>".
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</p>
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<p>The option "-protocols" of the ff* tools will display the list of
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supported protocols.
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</p>
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<p>All protocols accept the following options:
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</p>
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<dl compact="compact">
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<dt><span><samp>rw_timeout</samp></span></dt>
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<dd><p>Maximum time to wait for (network) read/write operations to complete,
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in microseconds.
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</p></dd>
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</dl>
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<p>A description of the currently available protocols follows.
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</p>
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<a name="amqp"></a>
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<h3 class="section">3.1 amqp<span class="pull-right"><a class="anchor hidden-xs" href="#amqp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-amqp" aria-hidden="true">TOC</a></span></h3>
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<p>Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
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publish-subscribe communication protocol.
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</p>
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<p>FFmpeg must be compiled with –enable-librabbitmq to support AMQP. A separate
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AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
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</p>
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<p>After starting the broker, an FFmpeg client may stream data to the broker using
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the command:
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</p>
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<div class="example">
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<pre class="example">ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]
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</pre></div>
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<p>Where hostname and port (default is 5672) is the address of the broker. The
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client may also set a user/password for authentication. The default for both
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fields is "guest". Name of virtual host on broker can be set with vhost. The
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default value is "/".
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</p>
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<p>Muliple subscribers may stream from the broker using the command:
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</p><div class="example">
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<pre class="example">ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]
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</pre></div>
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<p>In RabbitMQ all data published to the broker flows through a specific exchange,
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and each subscribing client has an assigned queue/buffer. When a packet arrives
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at an exchange, it may be copied to a client’s queue depending on the exchange
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and routing_key fields.
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</p>
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<p>The following options are supported:
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</p>
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<dl compact="compact">
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<dt><span><samp>exchange</samp></span></dt>
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<dd><p>Sets the exchange to use on the broker. RabbitMQ has several predefined
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exchanges: "amq.direct" is the default exchange, where the publisher and
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subscriber must have a matching routing_key; "amq.fanout" is the same as a
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broadcast operation (i.e. the data is forwarded to all queues on the fanout
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exchange independent of the routing_key); and "amq.topic" is similar to
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"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
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documentation).
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</p>
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</dd>
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<dt><span><samp>routing_key</samp></span></dt>
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<dd><p>Sets the routing key. The default value is "amqp". The routing key is used on
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the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
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to the queue of a subscriber.
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</p>
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</dd>
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<dt><span><samp>pkt_size</samp></span></dt>
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<dd><p>Maximum size of each packet sent/received to the broker. Default is 131072.
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Minimum is 4096 and max is any large value (representable by an int). When
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receiving packets, this sets an internal buffer size in FFmpeg. It should be
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equal to or greater than the size of the published packets to the broker. Otherwise
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the received message may be truncated causing decoding errors.
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</p>
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</dd>
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<dt><span><samp>connection_timeout</samp></span></dt>
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<dd><p>The timeout in seconds during the initial connection to the broker. The
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default value is rw_timeout, or 5 seconds if rw_timeout is not set.
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</p>
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</dd>
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<dt><span><samp>delivery_mode <var>mode</var></samp></span></dt>
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<dd><p>Sets the delivery mode of each message sent to broker.
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The following values are accepted:
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</p><dl compact="compact">
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<dt><span>‘<samp>persistent</samp>’</span></dt>
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<dd><p>Delivery mode set to "persistent" (2). This is the default value.
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Messages may be written to the broker’s disk depending on its setup.
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</p>
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</dd>
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<dt><span>‘<samp>non-persistent</samp>’</span></dt>
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<dd><p>Delivery mode set to "non-persistent" (1).
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Messages will stay in broker’s memory unless the broker is under memory
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pressure.
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</p>
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</dd>
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</dl>
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</dd>
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</dl>
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<a name="async"></a>
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<h3 class="section">3.2 async<span class="pull-right"><a class="anchor hidden-xs" href="#async" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-async" aria-hidden="true">TOC</a></span></h3>
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<p>Asynchronous data filling wrapper for input stream.
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</p>
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<p>Fill data in a background thread, to decouple I/O operation from demux thread.
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</p>
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<div class="example">
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<pre class="example">async:<var>URL</var>
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async:http://host/resource
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async:cache:http://host/resource
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</pre></div>
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<a name="bluray"></a>
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<h3 class="section">3.3 bluray<span class="pull-right"><a class="anchor hidden-xs" href="#bluray" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-bluray" aria-hidden="true">TOC</a></span></h3>
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<p>Read BluRay playlist.
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</p>
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<p>The accepted options are:
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</p><dl compact="compact">
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<dt><span><samp>angle</samp></span></dt>
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<dd><p>BluRay angle
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</p>
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</dd>
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<dt><span><samp>chapter</samp></span></dt>
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<dd><p>Start chapter (1...N)
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</p>
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</dd>
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<dt><span><samp>playlist</samp></span></dt>
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<dd><p>Playlist to read (BDMV/PLAYLIST/?????.mpls)
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</p>
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</dd>
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</dl>
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<p>Examples:
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</p>
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<p>Read longest playlist from BluRay mounted to /mnt/bluray:
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</p><div class="example">
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<pre class="example">bluray:/mnt/bluray
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</pre></div>
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<p>Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
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</p><div class="example">
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<pre class="example">-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
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</pre></div>
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<a name="cache"></a>
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||
<h3 class="section">3.4 cache<span class="pull-right"><a class="anchor hidden-xs" href="#cache" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-cache" aria-hidden="true">TOC</a></span></h3>
|
||
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<p>Caching wrapper for input stream.
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</p>
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||
<p>Cache the input stream to temporary file. It brings seeking capability to live streams.
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||
</p>
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||
<p>The accepted options are:
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||
</p><dl compact="compact">
|
||
<dt><span><samp>read_ahead_limit</samp></span></dt>
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||
<dd><p>Amount in bytes that may be read ahead when seeking isn’t supported. Range is -1 to INT_MAX.
|
||
-1 for unlimited. Default is 65536.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>URL Syntax is
|
||
</p><div class="example">
|
||
<pre class="example">cache:<var>URL</var>
|
||
</pre></div>
|
||
|
||
<a name="concat"></a>
|
||
<h3 class="section">3.5 concat<span class="pull-right"><a class="anchor hidden-xs" href="#concat" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-concat" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Physical concatenation protocol.
|
||
</p>
|
||
<p>Read and seek from many resources in sequence as if they were
|
||
a unique resource.
|
||
</p>
|
||
<p>A URL accepted by this protocol has the syntax:
|
||
</p><div class="example">
|
||
<pre class="example">concat:<var>URL1</var>|<var>URL2</var>|...|<var>URLN</var>
|
||
</pre></div>
|
||
|
||
<p>where <var>URL1</var>, <var>URL2</var>, ..., <var>URLN</var> are the urls of the
|
||
resource to be concatenated, each one possibly specifying a distinct
|
||
protocol.
|
||
</p>
|
||
<p>For example to read a sequence of files <samp>split1.mpeg</samp>,
|
||
<samp>split2.mpeg</samp>, <samp>split3.mpeg</samp> with <code>ffplay</code> use the
|
||
command:
|
||
</p><div class="example">
|
||
<pre class="example">ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
|
||
</pre></div>
|
||
|
||
<p>Note that you may need to escape the character "|" which is special for
|
||
many shells.
|
||
</p>
|
||
<a name="concatf"></a>
|
||
<h3 class="section">3.6 concatf<span class="pull-right"><a class="anchor hidden-xs" href="#concatf" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-concatf" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Physical concatenation protocol using a line break delimited list of
|
||
resources.
|
||
</p>
|
||
<p>Read and seek from many resources in sequence as if they were
|
||
a unique resource.
|
||
</p>
|
||
<p>A URL accepted by this protocol has the syntax:
|
||
</p><div class="example">
|
||
<pre class="example">concatf:<var>URL</var>
|
||
</pre></div>
|
||
|
||
<p>where <var>URL</var> is the url containing a line break delimited list of
|
||
resources to be concatenated, each one possibly specifying a distinct
|
||
protocol. Special characters must be escaped with backslash or single
|
||
quotes. See <a data-manual="ffmpeg-utils" href="ffmpeg-utils.html#quoting_005fand_005fescaping">(ffmpeg-utils)the "Quoting and escaping"
|
||
section in the ffmpeg-utils(1) manual</a>.
|
||
</p>
|
||
<p>For example to read a sequence of files <samp>split1.mpeg</samp>,
|
||
<samp>split2.mpeg</samp>, <samp>split3.mpeg</samp> listed in separate lines within
|
||
a file <samp>split.txt</samp> with <code>ffplay</code> use the command:
|
||
</p><div class="example">
|
||
<pre class="example">ffplay concatf:split.txt
|
||
</pre></div>
|
||
<p>Where <samp>split.txt</samp> contains the lines:
|
||
</p><div class="example">
|
||
<pre class="example">split1.mpeg
|
||
split2.mpeg
|
||
split3.mpeg
|
||
</pre></div>
|
||
|
||
<a name="crypto"></a>
|
||
<h3 class="section">3.7 crypto<span class="pull-right"><a class="anchor hidden-xs" href="#crypto" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-crypto" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>AES-encrypted stream reading protocol.
|
||
</p>
|
||
<p>The accepted options are:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>key</samp></span></dt>
|
||
<dd><p>Set the AES decryption key binary block from given hexadecimal representation.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>iv</samp></span></dt>
|
||
<dd><p>Set the AES decryption initialization vector binary block from given hexadecimal representation.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Accepted URL formats:
|
||
</p><div class="example">
|
||
<pre class="example">crypto:<var>URL</var>
|
||
crypto+<var>URL</var>
|
||
</pre></div>
|
||
|
||
<a name="data"></a>
|
||
<h3 class="section">3.8 data<span class="pull-right"><a class="anchor hidden-xs" href="#data" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-data" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Data in-line in the URI. See <a href="http://en.wikipedia.org/wiki/Data_URI_scheme">http://en.wikipedia.org/wiki/Data_URI_scheme</a>.
|
||
</p>
|
||
<p>For example, to convert a GIF file given inline with <code>ffmpeg</code>:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
|
||
</pre></div>
|
||
|
||
<a name="fd"></a>
|
||
<h3 class="section">3.9 fd<span class="pull-right"><a class="anchor hidden-xs" href="#fd" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-fd" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>File descriptor access protocol.
|
||
</p>
|
||
<p>The accepted syntax is:
|
||
</p><div class="example">
|
||
<pre class="example">fd: -fd <var>file_descriptor</var>
|
||
</pre></div>
|
||
|
||
<p>If <samp>fd</samp> is not specified, by default the stdout file descriptor will be
|
||
used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has
|
||
seek support if it corresponding to a regular file. fd protocol doesn’t support
|
||
pass file descriptor via URL for security.
|
||
</p>
|
||
<p>This protocol accepts the following options:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>blocksize</samp></span></dt>
|
||
<dd><p>Set I/O operation maximum block size, in bytes. Default value is
|
||
<code>INT_MAX</code>, which results in not limiting the requested block size.
|
||
Setting this value reasonably low improves user termination request reaction
|
||
time, which is valuable if data transmission is slow.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>fd</samp></span></dt>
|
||
<dd><p>Set file descriptor.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="file"></a>
|
||
<h3 class="section">3.10 file<span class="pull-right"><a class="anchor hidden-xs" href="#file" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-file" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>File access protocol.
|
||
</p>
|
||
<p>Read from or write to a file.
|
||
</p>
|
||
<p>A file URL can have the form:
|
||
</p><div class="example">
|
||
<pre class="example">file:<var>filename</var>
|
||
</pre></div>
|
||
|
||
<p>where <var>filename</var> is the path of the file to read.
|
||
</p>
|
||
<p>An URL that does not have a protocol prefix will be assumed to be a
|
||
file URL. Depending on the build, an URL that looks like a Windows
|
||
path with the drive letter at the beginning will also be assumed to be
|
||
a file URL (usually not the case in builds for unix-like systems).
|
||
</p>
|
||
<p>For example to read from a file <samp>input.mpeg</samp> with <code>ffmpeg</code>
|
||
use the command:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -i file:input.mpeg output.mpeg
|
||
</pre></div>
|
||
|
||
<p>This protocol accepts the following options:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>truncate</samp></span></dt>
|
||
<dd><p>Truncate existing files on write, if set to 1. A value of 0 prevents
|
||
truncating. Default value is 1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>blocksize</samp></span></dt>
|
||
<dd><p>Set I/O operation maximum block size, in bytes. Default value is
|
||
<code>INT_MAX</code>, which results in not limiting the requested block size.
|
||
Setting this value reasonably low improves user termination request reaction
|
||
time, which is valuable for files on slow medium.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>follow</samp></span></dt>
|
||
<dd><p>If set to 1, the protocol will retry reading at the end of the file, allowing
|
||
reading files that still are being written. In order for this to terminate,
|
||
you either need to use the rw_timeout option, or use the interrupt callback
|
||
(for API users).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>seekable</samp></span></dt>
|
||
<dd><p>Controls if seekability is advertised on the file. 0 means non-seekable, -1
|
||
means auto (seekable for normal files, non-seekable for named pipes).
|
||
</p>
|
||
<p>Many demuxers handle seekable and non-seekable resources differently,
|
||
overriding this might speed up opening certain files at the cost of losing some
|
||
features (e.g. accurate seeking).
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="ftp"></a>
|
||
<h3 class="section">3.11 ftp<span class="pull-right"><a class="anchor hidden-xs" href="#ftp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-ftp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>FTP (File Transfer Protocol).
|
||
</p>
|
||
<p>Read from or write to remote resources using FTP protocol.
|
||
</p>
|
||
<p>Following syntax is required.
|
||
</p><div class="example">
|
||
<pre class="example">ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
|
||
</pre></div>
|
||
|
||
<p>This protocol accepts the following options.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>timeout</samp></span></dt>
|
||
<dd><p>Set timeout in microseconds of socket I/O operations used by the underlying low level
|
||
operation. By default it is set to -1, which means that the timeout is
|
||
not specified.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ftp-user</samp></span></dt>
|
||
<dd><p>Set a user to be used for authenticating to the FTP server. This is overridden by the
|
||
user in the FTP URL.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ftp-password</samp></span></dt>
|
||
<dd><p>Set a password to be used for authenticating to the FTP server. This is overridden by
|
||
the password in the FTP URL, or by <samp>ftp-anonymous-password</samp> if no user is set.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ftp-anonymous-password</samp></span></dt>
|
||
<dd><p>Password used when login as anonymous user. Typically an e-mail address
|
||
should be used.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ftp-write-seekable</samp></span></dt>
|
||
<dd><p>Control seekability of connection during encoding. If set to 1 the
|
||
resource is supposed to be seekable, if set to 0 it is assumed not
|
||
to be seekable. Default value is 0.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>NOTE: Protocol can be used as output, but it is recommended to not do
|
||
it, unless special care is taken (tests, customized server configuration
|
||
etc.). Different FTP servers behave in different way during seek
|
||
operation. ff* tools may produce incomplete content due to server limitations.
|
||
</p>
|
||
<a name="gopher"></a>
|
||
<h3 class="section">3.12 gopher<span class="pull-right"><a class="anchor hidden-xs" href="#gopher" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-gopher" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Gopher protocol.
|
||
</p>
|
||
<a name="gophers"></a>
|
||
<h3 class="section">3.13 gophers<span class="pull-right"><a class="anchor hidden-xs" href="#gophers" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-gophers" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Gophers protocol.
|
||
</p>
|
||
<p>The Gopher protocol with TLS encapsulation.
|
||
</p>
|
||
<a name="hls"></a>
|
||
<h3 class="section">3.14 hls<span class="pull-right"><a class="anchor hidden-xs" href="#hls" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-hls" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Read Apple HTTP Live Streaming compliant segmented stream as
|
||
a uniform one. The M3U8 playlists describing the segments can be
|
||
remote HTTP resources or local files, accessed using the standard
|
||
file protocol.
|
||
The nested protocol is declared by specifying
|
||
"+<var>proto</var>" after the hls URI scheme name, where <var>proto</var>
|
||
is either "file" or "http".
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">hls+http://host/path/to/remote/resource.m3u8
|
||
hls+file://path/to/local/resource.m3u8
|
||
</pre></div>
|
||
|
||
<p>Using this protocol is discouraged - the hls demuxer should work
|
||
just as well (if not, please report the issues) and is more complete.
|
||
To use the hls demuxer instead, simply use the direct URLs to the
|
||
m3u8 files.
|
||
</p>
|
||
<a name="http"></a>
|
||
<h3 class="section">3.15 http<span class="pull-right"><a class="anchor hidden-xs" href="#http" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-http" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>HTTP (Hyper Text Transfer Protocol).
|
||
</p>
|
||
<p>This protocol accepts the following options:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>seekable</samp></span></dt>
|
||
<dd><p>Control seekability of connection. If set to 1 the resource is
|
||
supposed to be seekable, if set to 0 it is assumed not to be seekable,
|
||
if set to -1 it will try to autodetect if it is seekable. Default
|
||
value is -1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>chunked_post</samp></span></dt>
|
||
<dd><p>If set to 1 use chunked Transfer-Encoding for posts, default is 1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>content_type</samp></span></dt>
|
||
<dd><p>Set a specific content type for the POST messages or for listen mode.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>http_proxy</samp></span></dt>
|
||
<dd><p>set HTTP proxy to tunnel through e.g. http://example.com:1234
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>headers</samp></span></dt>
|
||
<dd><p>Set custom HTTP headers, can override built in default headers. The
|
||
value must be a string encoding the headers.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>multiple_requests</samp></span></dt>
|
||
<dd><p>Use persistent connections if set to 1, default is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>post_data</samp></span></dt>
|
||
<dd><p>Set custom HTTP post data.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>referer</samp></span></dt>
|
||
<dd><p>Set the Referer header. Include ’Referer: URL’ header in HTTP request.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>user_agent</samp></span></dt>
|
||
<dd><p>Override the User-Agent header. If not specified the protocol will use a
|
||
string describing the libavformat build. ("Lavf/<version>")
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reconnect_at_eof</samp></span></dt>
|
||
<dd><p>If set then eof is treated like an error and causes reconnection, this is useful
|
||
for live / endless streams.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reconnect_streamed</samp></span></dt>
|
||
<dd><p>If set then even streamed/non seekable streams will be reconnected on errors.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reconnect_on_network_error</samp></span></dt>
|
||
<dd><p>Reconnect automatically in case of TCP/TLS errors during connect.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reconnect_on_http_error</samp></span></dt>
|
||
<dd><p>A comma separated list of HTTP status codes to reconnect on. The list can
|
||
include specific status codes (e.g. ’503’) or the strings ’4xx’ / ’5xx’.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reconnect_delay_max</samp></span></dt>
|
||
<dd><p>Sets the maximum delay in seconds after which to give up reconnecting
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>mime_type</samp></span></dt>
|
||
<dd><p>Export the MIME type.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>http_version</samp></span></dt>
|
||
<dd><p>Exports the HTTP response version number. Usually "1.0" or "1.1".
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>icy</samp></span></dt>
|
||
<dd><p>If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
|
||
supports this, the metadata has to be retrieved by the application by reading
|
||
the <samp>icy_metadata_headers</samp> and <samp>icy_metadata_packet</samp> options.
|
||
The default is 1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>icy_metadata_headers</samp></span></dt>
|
||
<dd><p>If the server supports ICY metadata, this contains the ICY-specific HTTP reply
|
||
headers, separated by newline characters.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>icy_metadata_packet</samp></span></dt>
|
||
<dd><p>If the server supports ICY metadata, and <samp>icy</samp> was set to 1, this
|
||
contains the last non-empty metadata packet sent by the server. It should be
|
||
polled in regular intervals by applications interested in mid-stream metadata
|
||
updates.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>cookies</samp></span></dt>
|
||
<dd><p>Set the cookies to be sent in future requests. The format of each cookie is the
|
||
same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
|
||
delimited by a newline character.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>offset</samp></span></dt>
|
||
<dd><p>Set initial byte offset.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>end_offset</samp></span></dt>
|
||
<dd><p>Try to limit the request to bytes preceding this offset.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>method</samp></span></dt>
|
||
<dd><p>When used as a client option it sets the HTTP method for the request.
|
||
</p>
|
||
<p>When used as a server option it sets the HTTP method that is going to be
|
||
expected from the client(s).
|
||
If the expected and the received HTTP method do not match the client will
|
||
be given a Bad Request response.
|
||
When unset the HTTP method is not checked for now. This will be replaced by
|
||
autodetection in the future.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>listen</samp></span></dt>
|
||
<dd><p>If set to 1 enables experimental HTTP server. This can be used to send data when
|
||
used as an output option, or read data from a client with HTTP POST when used as
|
||
an input option.
|
||
If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
|
||
in ffmpeg.c and thus must not be used as a command line option.
|
||
</p><div class="example">
|
||
<pre class="example"># Server side (sending):
|
||
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<var>server</var>:<var>port</var>
|
||
|
||
# Client side (receiving):
|
||
ffmpeg -i http://<var>server</var>:<var>port</var> -c copy somefile.ogg
|
||
|
||
# Client can also be done with wget:
|
||
wget http://<var>server</var>:<var>port</var> -O somefile.ogg
|
||
|
||
# Server side (receiving):
|
||
ffmpeg -listen 1 -i http://<var>server</var>:<var>port</var> -c copy somefile.ogg
|
||
|
||
# Client side (sending):
|
||
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<var>server</var>:<var>port</var>
|
||
|
||
# Client can also be done with wget:
|
||
wget --post-file=somefile.ogg http://<var>server</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
</dd>
|
||
<dt><span><samp>send_expect_100</samp></span></dt>
|
||
<dd><p>Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
|
||
to 0 it won’t, if set to -1 it will try to send if it is applicable. Default
|
||
value is -1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>auth_type</samp></span></dt>
|
||
<dd>
|
||
<p>Set HTTP authentication type. No option for Digest, since this method requires
|
||
getting nonce parameters from the server first and can’t be used straight away like
|
||
Basic.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>none</samp></span></dt>
|
||
<dd><p>Choose the HTTP authentication type automatically. This is the default.
|
||
</p></dd>
|
||
<dt><span><samp>basic</samp></span></dt>
|
||
<dd>
|
||
<p>Choose the HTTP basic authentication.
|
||
</p>
|
||
<p>Basic authentication sends a Base64-encoded string that contains a user name and password
|
||
for the client. Base64 is not a form of encryption and should be considered the same as
|
||
sending the user name and password in clear text (Base64 is a reversible encoding).
|
||
If a resource needs to be protected, strongly consider using an authentication scheme
|
||
other than basic authentication. HTTPS/TLS should be used with basic authentication.
|
||
Without these additional security enhancements, basic authentication should not be used
|
||
to protect sensitive or valuable information.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
</dd>
|
||
</dl>
|
||
|
||
<a name="HTTP-Cookies"></a>
|
||
<h4 class="subsection">3.15.1 HTTP Cookies<span class="pull-right"><a class="anchor hidden-xs" href="#HTTP-Cookies" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-HTTP-Cookies" aria-hidden="true">TOC</a></span></h4>
|
||
|
||
<p>Some HTTP requests will be denied unless cookie values are passed in with the
|
||
request. The <samp>cookies</samp> option allows these cookies to be specified. At
|
||
the very least, each cookie must specify a value along with a path and domain.
|
||
HTTP requests that match both the domain and path will automatically include the
|
||
cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
|
||
by a newline.
|
||
</p>
|
||
<p>The required syntax to play a stream specifying a cookie is:
|
||
</p><div class="example">
|
||
<pre class="example">ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
|
||
</pre></div>
|
||
|
||
<a name="Icecast"></a>
|
||
<h3 class="section">3.16 Icecast<span class="pull-right"><a class="anchor hidden-xs" href="#Icecast" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Icecast" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Icecast protocol (stream to Icecast servers)
|
||
</p>
|
||
<p>This protocol accepts the following options:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>ice_genre</samp></span></dt>
|
||
<dd><p>Set the stream genre.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ice_name</samp></span></dt>
|
||
<dd><p>Set the stream name.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ice_description</samp></span></dt>
|
||
<dd><p>Set the stream description.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ice_url</samp></span></dt>
|
||
<dd><p>Set the stream website URL.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ice_public</samp></span></dt>
|
||
<dd><p>Set if the stream should be public.
|
||
The default is 0 (not public).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>user_agent</samp></span></dt>
|
||
<dd><p>Override the User-Agent header. If not specified a string of the form
|
||
"Lavf/<version>" will be used.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>password</samp></span></dt>
|
||
<dd><p>Set the Icecast mountpoint password.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>content_type</samp></span></dt>
|
||
<dd><p>Set the stream content type. This must be set if it is different from
|
||
audio/mpeg.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>legacy_icecast</samp></span></dt>
|
||
<dd><p>This enables support for Icecast versions < 2.4.0, that do not support the
|
||
HTTP PUT method but the SOURCE method.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tls</samp></span></dt>
|
||
<dd><p>Establish a TLS (HTTPS) connection to Icecast.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<div class="example">
|
||
<pre class="example">icecast://[<var>username</var>[:<var>password</var>]@]<var>server</var>:<var>port</var>/<var>mountpoint</var>
|
||
</pre></div>
|
||
|
||
<a name="ipfs"></a>
|
||
<h3 class="section">3.17 ipfs<span class="pull-right"><a class="anchor hidden-xs" href="#ipfs" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-ipfs" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>InterPlanetary File System (IPFS) protocol support. One can access files stored
|
||
on the IPFS network through so-called gateways. These are http(s) endpoints.
|
||
This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent
|
||
to such a gateway. Users can (and should) host their own node which means this
|
||
protocol will use one’s local gateway to access files on the IPFS network.
|
||
</p>
|
||
<p>This protocol accepts the following options:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>gateway</samp></span></dt>
|
||
<dd><p>Defines the gateway to use. When not set, the protocol will first try
|
||
locating the local gateway by looking at <code>$IPFS_GATEWAY</code>, <code>$IPFS_PATH</code>
|
||
and <code>$HOME/.ipfs/</code>, in that order.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>One can use this protocol in 2 ways. Using IPFS:
|
||
</p><div class="example">
|
||
<pre class="example">ffplay ipfs://<hash>
|
||
</pre></div>
|
||
|
||
<p>Or the IPNS protocol (IPNS is mutable IPFS):
|
||
</p><div class="example">
|
||
<pre class="example">ffplay ipns://<hash>
|
||
</pre></div>
|
||
|
||
<a name="mmst"></a>
|
||
<h3 class="section">3.18 mmst<span class="pull-right"><a class="anchor hidden-xs" href="#mmst" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mmst" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>MMS (Microsoft Media Server) protocol over TCP.
|
||
</p>
|
||
<a name="mmsh"></a>
|
||
<h3 class="section">3.19 mmsh<span class="pull-right"><a class="anchor hidden-xs" href="#mmsh" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mmsh" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>MMS (Microsoft Media Server) protocol over HTTP.
|
||
</p>
|
||
<p>The required syntax is:
|
||
</p><div class="example">
|
||
<pre class="example">mmsh://<var>server</var>[:<var>port</var>][/<var>app</var>][/<var>playpath</var>]
|
||
</pre></div>
|
||
|
||
<a name="md5"></a>
|
||
<h3 class="section">3.20 md5<span class="pull-right"><a class="anchor hidden-xs" href="#md5" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-md5" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>MD5 output protocol.
|
||
</p>
|
||
<p>Computes the MD5 hash of the data to be written, and on close writes
|
||
this to the designated output or stdout if none is specified. It can
|
||
be used to test muxers without writing an actual file.
|
||
</p>
|
||
<p>Some examples follow.
|
||
</p><div class="example">
|
||
<pre class="example"># Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
|
||
ffmpeg -i input.flv -f avi -y md5:output.avi.md5
|
||
|
||
# Write the MD5 hash of the encoded AVI file to stdout.
|
||
ffmpeg -i input.flv -f avi -y md5:
|
||
</pre></div>
|
||
|
||
<p>Note that some formats (typically MOV) require the output protocol to
|
||
be seekable, so they will fail with the MD5 output protocol.
|
||
</p>
|
||
<a name="pipe"></a>
|
||
<h3 class="section">3.21 pipe<span class="pull-right"><a class="anchor hidden-xs" href="#pipe" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-pipe" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>UNIX pipe access protocol.
|
||
</p>
|
||
<p>Read and write from UNIX pipes.
|
||
</p>
|
||
<p>The accepted syntax is:
|
||
</p><div class="example">
|
||
<pre class="example">pipe:[<var>number</var>]
|
||
</pre></div>
|
||
|
||
<p>If <samp>fd</samp> isn’t specified, <var>number</var> is the number corresponding to the file descriptor of the
|
||
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If <var>number</var>
|
||
is not specified, by default the stdout file descriptor will be used
|
||
for writing, stdin for reading.
|
||
</p>
|
||
<p>For example to read from stdin with <code>ffmpeg</code>:
|
||
</p><div class="example">
|
||
<pre class="example">cat test.wav | ffmpeg -i pipe:0
|
||
# ...this is the same as...
|
||
cat test.wav | ffmpeg -i pipe:
|
||
</pre></div>
|
||
|
||
<p>For writing to stdout with <code>ffmpeg</code>:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
|
||
# ...this is the same as...
|
||
ffmpeg -i test.wav -f avi pipe: | cat > test.avi
|
||
</pre></div>
|
||
|
||
<p>This protocol accepts the following options:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>blocksize</samp></span></dt>
|
||
<dd><p>Set I/O operation maximum block size, in bytes. Default value is
|
||
<code>INT_MAX</code>, which results in not limiting the requested block size.
|
||
Setting this value reasonably low improves user termination request reaction
|
||
time, which is valuable if data transmission is slow.
|
||
</p></dd>
|
||
<dt><span><samp>fd</samp></span></dt>
|
||
<dd><p>Set file descriptor.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Note that some formats (typically MOV), require the output protocol to
|
||
be seekable, so they will fail with the pipe output protocol.
|
||
</p>
|
||
<a name="prompeg"></a>
|
||
<h3 class="section">3.22 prompeg<span class="pull-right"><a class="anchor hidden-xs" href="#prompeg" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-prompeg" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
|
||
</p>
|
||
<p>The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
|
||
for MPEG-2 Transport Streams sent over RTP.
|
||
</p>
|
||
<p>This protocol must be used in conjunction with the <code>rtp_mpegts</code> muxer and
|
||
the <code>rtp</code> protocol.
|
||
</p>
|
||
<p>The required syntax is:
|
||
</p><div class="example">
|
||
<pre class="example">-f rtp_mpegts -fec prompeg=<var>option</var>=<var>val</var>... rtp://<var>hostname</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
<p>The destination UDP ports are <code>port + 2</code> for the column FEC stream
|
||
and <code>port + 4</code> for the row FEC stream.
|
||
</p>
|
||
<p>This protocol accepts the following options:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>l=<var>n</var></samp></span></dt>
|
||
<dd><p>The number of columns (4-20, LxD <= 100)
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>d=<var>n</var></samp></span></dt>
|
||
<dd><p>The number of rows (4-20, LxD <= 100)
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>Example usage:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<var>hostname</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
<a name="rist"></a>
|
||
<h3 class="section">3.23 rist<span class="pull-right"><a class="anchor hidden-xs" href="#rist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rist" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Reliable Internet Streaming Transport protocol
|
||
</p>
|
||
<p>The accepted options are:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>rist_profile</samp></span></dt>
|
||
<dd><p>Supported values:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>simple</samp>’</span></dt>
|
||
<dt><span>‘<samp>main</samp>’</span></dt>
|
||
<dd><p>This one is default.
|
||
</p></dd>
|
||
<dt><span>‘<samp>advanced</samp>’</span></dt>
|
||
</dl>
|
||
|
||
</dd>
|
||
<dt><span><samp>buffer_size</samp></span></dt>
|
||
<dd><p>Set internal RIST buffer size in milliseconds for retransmission of data.
|
||
Default value is 0 which means the librist default (1 sec). Maximum value is 30
|
||
seconds.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>fifo_size</samp></span></dt>
|
||
<dd><p>Size of the librist receiver output fifo in number of packets. This must be a
|
||
power of 2.
|
||
Defaults to 8192 (vs the librist default of 1024).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>overrun_nonfatal=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Survive in case of librist fifo buffer overrun. Default value is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>pkt_size</samp></span></dt>
|
||
<dd><p>Set maximum packet size for sending data. 1316 by default.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>log_level</samp></span></dt>
|
||
<dd><p>Set loglevel for RIST logging messages. You only need to set this if you
|
||
explicitly want to enable debug level messages or packet loss simulation,
|
||
otherwise the regular loglevel is respected.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>secret</samp></span></dt>
|
||
<dd><p>Set override of encryption secret, by default is unset.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>encryption</samp></span></dt>
|
||
<dd><p>Set encryption type, by default is disabled.
|
||
Acceptable values are 128 and 256.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="rtmp"></a>
|
||
<h3 class="section">3.24 rtmp<span class="pull-right"><a class="anchor hidden-xs" href="#rtmp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtmp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-Time Messaging Protocol.
|
||
</p>
|
||
<p>The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
|
||
content across a TCP/IP network.
|
||
</p>
|
||
<p>The required syntax is:
|
||
</p><div class="example">
|
||
<pre class="example">rtmp://[<var>username</var>:<var>password</var>@]<var>server</var>[:<var>port</var>][/<var>app</var>][/<var>instance</var>][/<var>playpath</var>]
|
||
</pre></div>
|
||
|
||
<p>The accepted parameters are:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>username</samp></span></dt>
|
||
<dd><p>An optional username (mostly for publishing).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>password</samp></span></dt>
|
||
<dd><p>An optional password (mostly for publishing).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>server</samp></span></dt>
|
||
<dd><p>The address of the RTMP server.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>port</samp></span></dt>
|
||
<dd><p>The number of the TCP port to use (by default is 1935).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>app</samp></span></dt>
|
||
<dd><p>It is the name of the application to access. It usually corresponds to
|
||
the path where the application is installed on the RTMP server
|
||
(e.g. <samp>/ondemand/</samp>, <samp>/flash/live/</samp>, etc.). You can override
|
||
the value parsed from the URI through the <code>rtmp_app</code> option, too.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>playpath</samp></span></dt>
|
||
<dd><p>It is the path or name of the resource to play with reference to the
|
||
application specified in <var>app</var>, may be prefixed by "mp4:". You
|
||
can override the value parsed from the URI through the <code>rtmp_playpath</code>
|
||
option, too.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>listen</samp></span></dt>
|
||
<dd><p>Act as a server, listening for an incoming connection.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>timeout</samp></span></dt>
|
||
<dd><p>Maximum time to wait for the incoming connection. Implies listen.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Additionally, the following parameters can be set via command line options
|
||
(or in code via <code>AVOption</code>s):
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>rtmp_app</samp></span></dt>
|
||
<dd><p>Name of application to connect on the RTMP server. This option
|
||
overrides the parameter specified in the URI.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_buffer</samp></span></dt>
|
||
<dd><p>Set the client buffer time in milliseconds. The default is 3000.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_conn</samp></span></dt>
|
||
<dd><p>Extra arbitrary AMF connection parameters, parsed from a string,
|
||
e.g. like <code>B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0</code>.
|
||
Each value is prefixed by a single character denoting the type,
|
||
B for Boolean, N for number, S for string, O for object, or Z for null,
|
||
followed by a colon. For Booleans the data must be either 0 or 1 for
|
||
FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
|
||
1 to end or begin an object, respectively. Data items in subobjects may
|
||
be named, by prefixing the type with ’N’ and specifying the name before
|
||
the value (i.e. <code>NB:myFlag:1</code>). This option may be used multiple
|
||
times to construct arbitrary AMF sequences.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_enhanced_codecs</samp></span></dt>
|
||
<dd><p>Specify the list of codecs the client advertises to support in an
|
||
enhanced RTMP stream. This option should be set to a comma separated
|
||
list of fourcc values, like <code>hvc1,av01,vp09</code> for multiple codecs
|
||
or <code>hvc1</code> for only one codec. The specified list will be presented
|
||
in the "fourCcLive" property of the Connect Command Message.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_flashver</samp></span></dt>
|
||
<dd><p>Version of the Flash plugin used to run the SWF player. The default
|
||
is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
|
||
<libavformat version>).)
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_flush_interval</samp></span></dt>
|
||
<dd><p>Number of packets flushed in the same request (RTMPT only). The default
|
||
is 10.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_live</samp></span></dt>
|
||
<dd><p>Specify that the media is a live stream. No resuming or seeking in
|
||
live streams is possible. The default value is <code>any</code>, which means the
|
||
subscriber first tries to play the live stream specified in the
|
||
playpath. If a live stream of that name is not found, it plays the
|
||
recorded stream. The other possible values are <code>live</code> and
|
||
<code>recorded</code>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_pageurl</samp></span></dt>
|
||
<dd><p>URL of the web page in which the media was embedded. By default no
|
||
value will be sent.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_playpath</samp></span></dt>
|
||
<dd><p>Stream identifier to play or to publish. This option overrides the
|
||
parameter specified in the URI.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_subscribe</samp></span></dt>
|
||
<dd><p>Name of live stream to subscribe to. By default no value will be sent.
|
||
It is only sent if the option is specified or if rtmp_live
|
||
is set to live.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_swfhash</samp></span></dt>
|
||
<dd><p>SHA256 hash of the decompressed SWF file (32 bytes).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_swfsize</samp></span></dt>
|
||
<dd><p>Size of the decompressed SWF file, required for SWFVerification.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_swfurl</samp></span></dt>
|
||
<dd><p>URL of the SWF player for the media. By default no value will be sent.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_swfverify</samp></span></dt>
|
||
<dd><p>URL to player swf file, compute hash/size automatically.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtmp_tcurl</samp></span></dt>
|
||
<dd><p>URL of the target stream. Defaults to proto://host[:port]/app.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tcp_nodelay=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0.
|
||
</p>
|
||
<p><em>Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.</em>
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>For example to read with <code>ffplay</code> a multimedia resource named
|
||
"sample" from the application "vod" from an RTMP server "myserver":
|
||
</p><div class="example">
|
||
<pre class="example">ffplay rtmp://myserver/vod/sample
|
||
</pre></div>
|
||
|
||
<p>To publish to a password protected server, passing the playpath and
|
||
app names separately:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/
|
||
</pre></div>
|
||
|
||
<a name="rtmpe"></a>
|
||
<h3 class="section">3.25 rtmpe<span class="pull-right"><a class="anchor hidden-xs" href="#rtmpe" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtmpe" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Encrypted Real-Time Messaging Protocol.
|
||
</p>
|
||
<p>The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
|
||
streaming multimedia content within standard cryptographic primitives,
|
||
consisting of Diffie-Hellman key exchange and HMACSHA256, generating
|
||
a pair of RC4 keys.
|
||
</p>
|
||
<a name="rtmps"></a>
|
||
<h3 class="section">3.26 rtmps<span class="pull-right"><a class="anchor hidden-xs" href="#rtmps" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtmps" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-Time Messaging Protocol over a secure SSL connection.
|
||
</p>
|
||
<p>The Real-Time Messaging Protocol (RTMPS) is used for streaming
|
||
multimedia content across an encrypted connection.
|
||
</p>
|
||
<a name="rtmpt"></a>
|
||
<h3 class="section">3.27 rtmpt<span class="pull-right"><a class="anchor hidden-xs" href="#rtmpt" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtmpt" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-Time Messaging Protocol tunneled through HTTP.
|
||
</p>
|
||
<p>The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
|
||
for streaming multimedia content within HTTP requests to traverse
|
||
firewalls.
|
||
</p>
|
||
<a name="rtmpte"></a>
|
||
<h3 class="section">3.28 rtmpte<span class="pull-right"><a class="anchor hidden-xs" href="#rtmpte" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtmpte" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Encrypted Real-Time Messaging Protocol tunneled through HTTP.
|
||
</p>
|
||
<p>The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
|
||
is used for streaming multimedia content within HTTP requests to traverse
|
||
firewalls.
|
||
</p>
|
||
<a name="rtmpts"></a>
|
||
<h3 class="section">3.29 rtmpts<span class="pull-right"><a class="anchor hidden-xs" href="#rtmpts" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtmpts" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-Time Messaging Protocol tunneled through HTTPS.
|
||
</p>
|
||
<p>The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
|
||
for streaming multimedia content within HTTPS requests to traverse
|
||
firewalls.
|
||
</p>
|
||
<a name="libsmbclient"></a>
|
||
<h3 class="section">3.30 libsmbclient<span class="pull-right"><a class="anchor hidden-xs" href="#libsmbclient" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-libsmbclient" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>libsmbclient permits one to manipulate CIFS/SMB network resources.
|
||
</p>
|
||
<p>Following syntax is required.
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
|
||
</pre></div>
|
||
|
||
<p>This protocol accepts the following options.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>timeout</samp></span></dt>
|
||
<dd><p>Set timeout in milliseconds of socket I/O operations used by the underlying
|
||
low level operation. By default it is set to -1, which means that the timeout
|
||
is not specified.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>truncate</samp></span></dt>
|
||
<dd><p>Truncate existing files on write, if set to 1. A value of 0 prevents
|
||
truncating. Default value is 1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>workgroup</samp></span></dt>
|
||
<dd><p>Set the workgroup used for making connections. By default workgroup is not specified.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>For more information see: <a href="http://www.samba.org/">http://www.samba.org/</a>.
|
||
</p>
|
||
<a name="libssh"></a>
|
||
<h3 class="section">3.31 libssh<span class="pull-right"><a class="anchor hidden-xs" href="#libssh" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-libssh" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Secure File Transfer Protocol via libssh
|
||
</p>
|
||
<p>Read from or write to remote resources using SFTP protocol.
|
||
</p>
|
||
<p>Following syntax is required.
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
|
||
</pre></div>
|
||
|
||
<p>This protocol accepts the following options.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>timeout</samp></span></dt>
|
||
<dd><p>Set timeout of socket I/O operations used by the underlying low level
|
||
operation. By default it is set to -1, which means that the timeout
|
||
is not specified.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>truncate</samp></span></dt>
|
||
<dd><p>Truncate existing files on write, if set to 1. A value of 0 prevents
|
||
truncating. Default value is 1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>private_key</samp></span></dt>
|
||
<dd><p>Specify the path of the file containing private key to use during authorization.
|
||
By default libssh searches for keys in the <samp>~/.ssh/</samp> directory.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>Example: Play a file stored on remote server.
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
|
||
</pre></div>
|
||
|
||
<a name="librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte"></a>
|
||
<h3 class="section">3.32 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte<span class="pull-right"><a class="anchor hidden-xs" href="#librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-Time Messaging Protocol and its variants supported through
|
||
librtmp.
|
||
</p>
|
||
<p>Requires the presence of the librtmp headers and library during
|
||
configuration. You need to explicitly configure the build with
|
||
"–enable-librtmp". If enabled this will replace the native RTMP
|
||
protocol.
|
||
</p>
|
||
<p>This protocol provides most client functions and a few server
|
||
functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
|
||
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
|
||
variants of these encrypted types (RTMPTE, RTMPTS).
|
||
</p>
|
||
<p>The required syntax is:
|
||
</p><div class="example">
|
||
<pre class="example"><var>rtmp_proto</var>://<var>server</var>[:<var>port</var>][/<var>app</var>][/<var>playpath</var>] <var>options</var>
|
||
</pre></div>
|
||
|
||
<p>where <var>rtmp_proto</var> is one of the strings "rtmp", "rtmpt", "rtmpe",
|
||
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
|
||
<var>server</var>, <var>port</var>, <var>app</var> and <var>playpath</var> have the same
|
||
meaning as specified for the RTMP native protocol.
|
||
<var>options</var> contains a list of space-separated options of the form
|
||
<var>key</var>=<var>val</var>.
|
||
</p>
|
||
<p>See the librtmp manual page (man 3 librtmp) for more information.
|
||
</p>
|
||
<p>For example, to stream a file in real-time to an RTMP server using
|
||
<code>ffmpeg</code>:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
|
||
</pre></div>
|
||
|
||
<p>To play the same stream using <code>ffplay</code>:
|
||
</p><div class="example">
|
||
<pre class="example">ffplay "rtmp://myserver/live/mystream live=1"
|
||
</pre></div>
|
||
|
||
<a name="rtp"></a>
|
||
<h3 class="section">3.33 rtp<span class="pull-right"><a class="anchor hidden-xs" href="#rtp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-time Transport Protocol.
|
||
</p>
|
||
<p>The required syntax for an RTP URL is:
|
||
rtp://<var>hostname</var>[:<var>port</var>][?<var>option</var>=<var>val</var>...]
|
||
</p>
|
||
<p><var>port</var> specifies the RTP port to use.
|
||
</p>
|
||
<p>The following URL options are supported:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>ttl=<var>n</var></samp></span></dt>
|
||
<dd><p>Set the TTL (Time-To-Live) value (for multicast only).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtcpport=<var>n</var></samp></span></dt>
|
||
<dd><p>Set the remote RTCP port to <var>n</var>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>localrtpport=<var>n</var></samp></span></dt>
|
||
<dd><p>Set the local RTP port to <var>n</var>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>localrtcpport=<var>n</var>'</samp></span></dt>
|
||
<dd><p>Set the local RTCP port to <var>n</var>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>pkt_size=<var>n</var></samp></span></dt>
|
||
<dd><p>Set max packet size (in bytes) to <var>n</var>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>buffer_size=<var>size</var></samp></span></dt>
|
||
<dd><p>Set the maximum UDP socket buffer size in bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>connect=0|1</samp></span></dt>
|
||
<dd><p>Do a <code>connect()</code> on the UDP socket (if set to 1) or not (if set
|
||
to 0).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>sources=<var>ip</var>[,<var>ip</var>]</samp></span></dt>
|
||
<dd><p>List allowed source IP addresses.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>block=<var>ip</var>[,<var>ip</var>]</samp></span></dt>
|
||
<dd><p>List disallowed (blocked) source IP addresses.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>write_to_source=0|1</samp></span></dt>
|
||
<dd><p>Send packets to the source address of the latest received packet (if
|
||
set to 1) or to a default remote address (if set to 0).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>localport=<var>n</var></samp></span></dt>
|
||
<dd><p>Set the local RTP port to <var>n</var>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>localaddr=<var>addr</var></samp></span></dt>
|
||
<dd><p>Local IP address of a network interface used for sending packets or joining
|
||
multicast groups.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>timeout=<var>n</var></samp></span></dt>
|
||
<dd><p>Set timeout (in microseconds) of socket I/O operations to <var>n</var>.
|
||
</p>
|
||
<p>This is a deprecated option. Instead, <samp>localrtpport</samp> should be
|
||
used.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>Important notes:
|
||
</p>
|
||
<ol>
|
||
<li> If <samp>rtcpport</samp> is not set the RTCP port will be set to the RTP
|
||
port value plus 1.
|
||
|
||
</li><li> If <samp>localrtpport</samp> (the local RTP port) is not set any available
|
||
port will be used for the local RTP and RTCP ports.
|
||
|
||
</li><li> If <samp>localrtcpport</samp> (the local RTCP port) is not set it will be
|
||
set to the local RTP port value plus 1.
|
||
</li></ol>
|
||
|
||
<a name="rtsp"></a>
|
||
<h3 class="section">3.34 rtsp<span class="pull-right"><a class="anchor hidden-xs" href="#rtsp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-rtsp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Real-Time Streaming Protocol.
|
||
</p>
|
||
<p>RTSP is not technically a protocol handler in libavformat, it is a demuxer
|
||
and muxer. The demuxer supports both normal RTSP (with data transferred
|
||
over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
|
||
data transferred over RDT).
|
||
</p>
|
||
<p>The muxer can be used to send a stream using RTSP ANNOUNCE to a server
|
||
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock’s
|
||
<a href="https://github.com/revmischa/rtsp-server">RTSP server</a>).
|
||
</p>
|
||
<p>The required syntax for a RTSP url is:
|
||
</p><div class="example">
|
||
<pre class="example">rtsp://<var>hostname</var>[:<var>port</var>]/<var>path</var>
|
||
</pre></div>
|
||
|
||
<p>Options can be set on the <code>ffmpeg</code>/<code>ffplay</code> command
|
||
line, or set in code via <code>AVOption</code>s or in
|
||
<code>avformat_open_input</code>.
|
||
</p>
|
||
<a name="Muxer"></a>
|
||
<h4 class="subsection">3.34.1 Muxer<span class="pull-right"><a class="anchor hidden-xs" href="#Muxer" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Muxer" aria-hidden="true">TOC</a></span></h4>
|
||
<p>The following options are supported.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>rtsp_transport</samp></span></dt>
|
||
<dd><p>Set RTSP transport protocols.
|
||
</p>
|
||
<p>It accepts the following values:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>udp</samp>’</span></dt>
|
||
<dd><p>Use UDP as lower transport protocol.
|
||
</p>
|
||
</dd>
|
||
<dt><span>‘<samp>tcp</samp>’</span></dt>
|
||
<dd><p>Use TCP (interleaving within the RTSP control channel) as lower
|
||
transport protocol.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Default value is ‘<samp>0</samp>’.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtsp_flags</samp></span></dt>
|
||
<dd><p>Set RTSP flags.
|
||
</p>
|
||
<p>The following values are accepted:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>latm</samp>’</span></dt>
|
||
<dd><p>Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.
|
||
</p></dd>
|
||
<dt><span>‘<samp>rfc2190</samp>’</span></dt>
|
||
<dd><p>Use RFC 2190 packetization instead of RFC 4629 for H.263.
|
||
</p></dd>
|
||
<dt><span>‘<samp>skip_rtcp</samp>’</span></dt>
|
||
<dd><p>Don’t send RTCP sender reports.
|
||
</p></dd>
|
||
<dt><span>‘<samp>h264_mode0</samp>’</span></dt>
|
||
<dd><p>Use mode 0 for H.264 in RTP.
|
||
</p></dd>
|
||
<dt><span>‘<samp>send_bye</samp>’</span></dt>
|
||
<dd><p>Send RTCP BYE packets when finishing.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Default value is ‘<samp>0</samp>’.
|
||
</p>
|
||
|
||
</dd>
|
||
<dt><span><samp>min_port</samp></span></dt>
|
||
<dd><p>Set minimum local UDP port. Default value is 5000.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>max_port</samp></span></dt>
|
||
<dd><p>Set maximum local UDP port. Default value is 65000.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>buffer_size</samp></span></dt>
|
||
<dd><p>Set the maximum socket buffer size in bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>pkt_size</samp></span></dt>
|
||
<dd><p>Set max send packet size (in bytes). Default value is 1472.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="Demuxer"></a>
|
||
<h4 class="subsection">3.34.2 Demuxer<span class="pull-right"><a class="anchor hidden-xs" href="#Demuxer" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Demuxer" aria-hidden="true">TOC</a></span></h4>
|
||
<p>The following options are supported.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>initial_pause</samp></span></dt>
|
||
<dd><p>Do not start playing the stream immediately if set to 1. Default value
|
||
is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtsp_transport</samp></span></dt>
|
||
<dd><p>Set RTSP transport protocols.
|
||
</p>
|
||
<p>It accepts the following values:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>udp</samp>’</span></dt>
|
||
<dd><p>Use UDP as lower transport protocol.
|
||
</p>
|
||
</dd>
|
||
<dt><span>‘<samp>tcp</samp>’</span></dt>
|
||
<dd><p>Use TCP (interleaving within the RTSP control channel) as lower
|
||
transport protocol.
|
||
</p>
|
||
</dd>
|
||
<dt><span>‘<samp>udp_multicast</samp>’</span></dt>
|
||
<dd><p>Use UDP multicast as lower transport protocol.
|
||
</p>
|
||
</dd>
|
||
<dt><span>‘<samp>http</samp>’</span></dt>
|
||
<dd><p>Use HTTP tunneling as lower transport protocol, which is useful for
|
||
passing proxies.
|
||
</p>
|
||
</dd>
|
||
<dt><span>‘<samp>https</samp>’</span></dt>
|
||
<dd><p>Use HTTPs tunneling as lower transport protocol, which is useful for
|
||
passing proxies and widely used for security consideration.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Multiple lower transport protocols may be specified, in that case they are
|
||
tried one at a time (if the setup of one fails, the next one is tried).
|
||
For the muxer, only the ‘<samp>tcp</samp>’ and ‘<samp>udp</samp>’ options are supported.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rtsp_flags</samp></span></dt>
|
||
<dd><p>Set RTSP flags.
|
||
</p>
|
||
<p>The following values are accepted:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>filter_src</samp>’</span></dt>
|
||
<dd><p>Accept packets only from negotiated peer address and port.
|
||
</p></dd>
|
||
<dt><span>‘<samp>listen</samp>’</span></dt>
|
||
<dd><p>Act as a server, listening for an incoming connection.
|
||
</p></dd>
|
||
<dt><span>‘<samp>prefer_tcp</samp>’</span></dt>
|
||
<dd><p>Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
|
||
</p></dd>
|
||
<dt><span>‘<samp>satip_raw</samp>’</span></dt>
|
||
<dd><p>Export raw MPEG-TS stream instead of demuxing. The flag will simply write out
|
||
the raw stream, with the original PAT/PMT/PIDs intact.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Default value is ‘<samp>none</samp>’.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>allowed_media_types</samp></span></dt>
|
||
<dd><p>Set media types to accept from the server.
|
||
</p>
|
||
<p>The following flags are accepted:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>video</samp>’</span></dt>
|
||
<dt><span>‘<samp>audio</samp>’</span></dt>
|
||
<dt><span>‘<samp>data</samp>’</span></dt>
|
||
<dt><span>‘<samp>subtitle</samp>’</span></dt>
|
||
</dl>
|
||
|
||
<p>By default it accepts all media types.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>min_port</samp></span></dt>
|
||
<dd><p>Set minimum local UDP port. Default value is 5000.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>max_port</samp></span></dt>
|
||
<dd><p>Set maximum local UDP port. Default value is 65000.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>listen_timeout</samp></span></dt>
|
||
<dd><p>Set maximum timeout (in seconds) to establish an initial connection. Setting
|
||
<samp>listen_timeout</samp> > 0 sets <samp>rtsp_flags</samp> to ‘<samp>listen</samp>’. Default is -1
|
||
which means an infinite timeout when ‘<samp>listen</samp>’ mode is set.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reorder_queue_size</samp></span></dt>
|
||
<dd><p>Set number of packets to buffer for handling of reordered packets.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>timeout</samp></span></dt>
|
||
<dd><p>Set socket TCP I/O timeout in microseconds.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>user_agent</samp></span></dt>
|
||
<dd><p>Override User-Agent header. If not specified, it defaults to the
|
||
libavformat identifier string.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>buffer_size</samp></span></dt>
|
||
<dd><p>Set the maximum socket buffer size in bytes.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>When receiving data over UDP, the demuxer tries to reorder received packets
|
||
(since they may arrive out of order, or packets may get lost totally). This
|
||
can be disabled by setting the maximum demuxing delay to zero (via
|
||
the <code>max_delay</code> field of AVFormatContext).
|
||
</p>
|
||
<p>When watching multi-bitrate Real-RTSP streams with <code>ffplay</code>, the
|
||
streams to display can be chosen with <code>-vst</code> <var>n</var> and
|
||
<code>-ast</code> <var>n</var> for video and audio respectively, and can be switched
|
||
on the fly by pressing <code>v</code> and <code>a</code>.
|
||
</p>
|
||
<a name="Examples"></a>
|
||
<h4 class="subsection">3.34.3 Examples<span class="pull-right"><a class="anchor hidden-xs" href="#Examples" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Examples" aria-hidden="true">TOC</a></span></h4>
|
||
|
||
<p>The following examples all make use of the <code>ffplay</code> and
|
||
<code>ffmpeg</code> tools.
|
||
</p>
|
||
<ul>
|
||
<li> Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
|
||
<div class="example">
|
||
<pre class="example">ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
|
||
</pre></div>
|
||
|
||
</li><li> Watch a stream tunneled over HTTP:
|
||
<div class="example">
|
||
<pre class="example">ffplay -rtsp_transport http rtsp://server/video.mp4
|
||
</pre></div>
|
||
|
||
</li><li> Send a stream in realtime to a RTSP server, for others to watch:
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -re -i <var>input</var> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
|
||
</pre></div>
|
||
|
||
</li><li> Receive a stream in realtime:
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <var>output</var>
|
||
</pre></div>
|
||
</li></ul>
|
||
|
||
<a name="sap"></a>
|
||
<h3 class="section">3.35 sap<span class="pull-right"><a class="anchor hidden-xs" href="#sap" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-sap" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Session Announcement Protocol (RFC 2974). This is not technically a
|
||
protocol handler in libavformat, it is a muxer and demuxer.
|
||
It is used for signalling of RTP streams, by announcing the SDP for the
|
||
streams regularly on a separate port.
|
||
</p>
|
||
<a name="Muxer-1"></a>
|
||
<h4 class="subsection">3.35.1 Muxer<span class="pull-right"><a class="anchor hidden-xs" href="#Muxer-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Muxer-1" aria-hidden="true">TOC</a></span></h4>
|
||
|
||
<p>The syntax for a SAP url given to the muxer is:
|
||
</p><div class="example">
|
||
<pre class="example">sap://<var>destination</var>[:<var>port</var>][?<var>options</var>]
|
||
</pre></div>
|
||
|
||
<p>The RTP packets are sent to <var>destination</var> on port <var>port</var>,
|
||
or to port 5004 if no port is specified.
|
||
<var>options</var> is a <code>&</code>-separated list. The following options
|
||
are supported:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>announce_addr=<var>address</var></samp></span></dt>
|
||
<dd><p>Specify the destination IP address for sending the announcements to.
|
||
If omitted, the announcements are sent to the commonly used SAP
|
||
announcement multicast address 224.2.127.254 (sap.mcast.net), or
|
||
ff0e::2:7ffe if <var>destination</var> is an IPv6 address.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>announce_port=<var>port</var></samp></span></dt>
|
||
<dd><p>Specify the port to send the announcements on, defaults to
|
||
9875 if not specified.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ttl=<var>ttl</var></samp></span></dt>
|
||
<dd><p>Specify the time to live value for the announcements and RTP packets,
|
||
defaults to 255.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>same_port=<var>0|1</var></samp></span></dt>
|
||
<dd><p>If set to 1, send all RTP streams on the same port pair. If zero (the
|
||
default), all streams are sent on unique ports, with each stream on a
|
||
port 2 numbers higher than the previous.
|
||
VLC/Live555 requires this to be set to 1, to be able to receive the stream.
|
||
The RTP stack in libavformat for receiving requires all streams to be sent
|
||
on unique ports.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Example command lines follow.
|
||
</p>
|
||
<p>To broadcast a stream on the local subnet, for watching in VLC:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -re -i <var>input</var> -f sap sap://224.0.0.255?same_port=1
|
||
</pre></div>
|
||
|
||
<p>Similarly, for watching in <code>ffplay</code>:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -re -i <var>input</var> -f sap sap://224.0.0.255
|
||
</pre></div>
|
||
|
||
<p>And for watching in <code>ffplay</code>, over IPv6:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -re -i <var>input</var> -f sap sap://[ff0e::1:2:3:4]
|
||
</pre></div>
|
||
|
||
<a name="Demuxer-1"></a>
|
||
<h4 class="subsection">3.35.2 Demuxer<span class="pull-right"><a class="anchor hidden-xs" href="#Demuxer-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Demuxer-1" aria-hidden="true">TOC</a></span></h4>
|
||
|
||
<p>The syntax for a SAP url given to the demuxer is:
|
||
</p><div class="example">
|
||
<pre class="example">sap://[<var>address</var>][:<var>port</var>]
|
||
</pre></div>
|
||
|
||
<p><var>address</var> is the multicast address to listen for announcements on,
|
||
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. <var>port</var>
|
||
is the port that is listened on, 9875 if omitted.
|
||
</p>
|
||
<p>The demuxers listens for announcements on the given address and port.
|
||
Once an announcement is received, it tries to receive that particular stream.
|
||
</p>
|
||
<p>Example command lines follow.
|
||
</p>
|
||
<p>To play back the first stream announced on the normal SAP multicast address:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffplay sap://
|
||
</pre></div>
|
||
|
||
<p>To play back the first stream announced on one the default IPv6 SAP multicast address:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffplay sap://[ff0e::2:7ffe]
|
||
</pre></div>
|
||
|
||
<a name="sctp"></a>
|
||
<h3 class="section">3.36 sctp<span class="pull-right"><a class="anchor hidden-xs" href="#sctp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-sctp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Stream Control Transmission Protocol.
|
||
</p>
|
||
<p>The accepted URL syntax is:
|
||
</p><div class="example">
|
||
<pre class="example">sctp://<var>host</var>:<var>port</var>[?<var>options</var>]
|
||
</pre></div>
|
||
|
||
<p>The protocol accepts the following options:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>listen</samp></span></dt>
|
||
<dd><p>If set to any value, listen for an incoming connection. Outgoing connection is done by default.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>max_streams</samp></span></dt>
|
||
<dd><p>Set the maximum number of streams. By default no limit is set.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="srt"></a>
|
||
<h3 class="section">3.37 srt<span class="pull-right"><a class="anchor hidden-xs" href="#srt" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-srt" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Haivision Secure Reliable Transport Protocol via libsrt.
|
||
</p>
|
||
<p>The supported syntax for a SRT URL is:
|
||
</p><div class="example">
|
||
<pre class="example">srt://<var>hostname</var>:<var>port</var>[?<var>options</var>]
|
||
</pre></div>
|
||
|
||
<p><var>options</var> contains a list of &-separated options of the form
|
||
<var>key</var>=<var>val</var>.
|
||
</p>
|
||
<p>or
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example"><var>options</var> srt://<var>hostname</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
<p><var>options</var> contains a list of ’-<var>key</var> <var>val</var>’
|
||
options.
|
||
</p>
|
||
<p>This protocol accepts the following options.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>connect_timeout=<var>milliseconds</var></samp></span></dt>
|
||
<dd><p>Connection timeout; SRT cannot connect for RTT > 1500 msec
|
||
(2 handshake exchanges) with the default connect timeout of
|
||
3 seconds. This option applies to the caller and rendezvous
|
||
connection modes. The connect timeout is 10 times the value
|
||
set for the rendezvous mode (which can be used as a
|
||
workaround for this connection problem with earlier versions).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ffs=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Flight Flag Size (Window Size), in bytes. FFS is actually an
|
||
internal parameter and you should set it to not less than
|
||
<samp>recv_buffer_size</samp> and <samp>mss</samp>. The default value
|
||
is relatively large, therefore unless you set a very large receiver buffer,
|
||
you do not need to change this option. Default value is 25600.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>inputbw=<var>bytes/seconds</var></samp></span></dt>
|
||
<dd><p>Sender nominal input rate, in bytes per seconds. Used along with
|
||
<samp>oheadbw</samp>, when <samp>maxbw</samp> is set to relative (0), to
|
||
calculate maximum sending rate when recovery packets are sent
|
||
along with the main media stream:
|
||
<samp>inputbw</samp> * (100 + <samp>oheadbw</samp>) / 100
|
||
if <samp>inputbw</samp> is not set while <samp>maxbw</samp> is set to
|
||
relative (0), the actual input rate is evaluated inside
|
||
the library. Default value is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>iptos=<var>tos</var></samp></span></dt>
|
||
<dd><p>IP Type of Service. Applies to sender only. Default value is 0xB8.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ipttl=<var>ttl</var></samp></span></dt>
|
||
<dd><p>IP Time To Live. Applies to sender only. Default value is 64.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>latency=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>Timestamp-based Packet Delivery Delay.
|
||
Used to absorb bursts of missed packet retransmissions.
|
||
This flag sets both <samp>rcvlatency</samp> and <samp>peerlatency</samp>
|
||
to the same value. Note that prior to version 1.3.0
|
||
this is the only flag to set the latency, however
|
||
this is effectively equivalent to setting <samp>peerlatency</samp>,
|
||
when side is sender and <samp>rcvlatency</samp>
|
||
when side is receiver, and the bidirectional stream
|
||
sending is not supported.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>listen_timeout=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>Set socket listen timeout.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>maxbw=<var>bytes/seconds</var></samp></span></dt>
|
||
<dd><p>Maximum sending bandwidth, in bytes per seconds.
|
||
-1 infinite (CSRTCC limit is 30mbps)
|
||
0 relative to input rate (see <samp>inputbw</samp>)
|
||
>0 absolute limit value
|
||
Default value is 0 (relative)
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>mode=<var>caller|listener|rendezvous</var></samp></span></dt>
|
||
<dd><p>Connection mode.
|
||
<samp>caller</samp> opens client connection.
|
||
<samp>listener</samp> starts server to listen for incoming connections.
|
||
<samp>rendezvous</samp> use Rendez-Vous connection mode.
|
||
Default value is caller.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>mss=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Maximum Segment Size, in bytes. Used for buffer allocation
|
||
and rate calculation using a packet counter assuming fully
|
||
filled packets. The smallest MSS between the peers is
|
||
used. This is 1500 by default in the overall internet.
|
||
This is the maximum size of the UDP packet and can be
|
||
only decreased, unless you have some unusual dedicated
|
||
network settings. Default value is 1500.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>nakreport=<var>1|0</var></samp></span></dt>
|
||
<dd><p>If set to 1, Receiver will send ‘UMSG_LOSSREPORT‘ messages
|
||
periodically until a lost packet is retransmitted or
|
||
intentionally dropped. Default value is 1.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>oheadbw=<var>percents</var></samp></span></dt>
|
||
<dd><p>Recovery bandwidth overhead above input rate, in percents.
|
||
See <samp>inputbw</samp>. Default value is 25%.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>passphrase=<var>string</var></samp></span></dt>
|
||
<dd><p>HaiCrypt Encryption/Decryption Passphrase string, length
|
||
from 10 to 79 characters. The passphrase is the shared
|
||
secret between the sender and the receiver. It is used
|
||
to generate the Key Encrypting Key using PBKDF2
|
||
(Password-Based Key Derivation Function). It is used
|
||
only if <samp>pbkeylen</samp> is non-zero. It is used on
|
||
the receiver only if the received data is encrypted.
|
||
The configured passphrase cannot be recovered (write-only).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>enforced_encryption=<var>1|0</var></samp></span></dt>
|
||
<dd><p>If true, both connection parties must have the same password
|
||
set (including empty, that is, with no encryption). If the
|
||
password doesn’t match or only one side is unencrypted,
|
||
the connection is rejected. Default is true.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>kmrefreshrate=<var>packets</var></samp></span></dt>
|
||
<dd><p>The number of packets to be transmitted after which the
|
||
encryption key is switched to a new key. Default is -1.
|
||
-1 means auto (0x1000000 in srt library). The range for
|
||
this option is integers in the 0 - <code>INT_MAX</code>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>kmpreannounce=<var>packets</var></samp></span></dt>
|
||
<dd><p>The interval between when a new encryption key is sent and
|
||
when switchover occurs. This value also applies to the
|
||
subsequent interval between when switchover occurs and
|
||
when the old encryption key is decommissioned. Default is -1.
|
||
-1 means auto (0x1000 in srt library). The range for
|
||
this option is integers in the 0 - <code>INT_MAX</code>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>snddropdelay=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>The sender’s extra delay before dropping packets. This delay is
|
||
added to the default drop delay time interval value.
|
||
</p>
|
||
<p>Special value -1: Do not drop packets on the sender at all.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>payload_size=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Sets the maximum declared size of a packet transferred
|
||
during the single call to the sending function in Live
|
||
mode. Use 0 if this value isn’t used (which is default in
|
||
file mode).
|
||
Default is -1 (automatic), which typically means MPEG-TS;
|
||
if you are going to use SRT
|
||
to send any different kind of payload, such as, for example,
|
||
wrapping a live stream in very small frames, then you can
|
||
use a bigger maximum frame size, though not greater than
|
||
1456 bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>pkt_size=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Alias for ‘<samp>payload_size</samp>’.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>peerlatency=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>The latency value (as described in <samp>rcvlatency</samp>) that is
|
||
set by the sender side as a minimum value for the receiver.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>pbkeylen=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Sender encryption key length, in bytes.
|
||
Only can be set to 0, 16, 24 and 32.
|
||
Enable sender encryption if not 0.
|
||
Not required on receiver (set to 0),
|
||
key size obtained from sender in HaiCrypt handshake.
|
||
Default value is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rcvlatency=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>The time that should elapse since the moment when the
|
||
packet was sent and the moment when it’s delivered to
|
||
the receiver application in the receiving function.
|
||
This time should be a buffer time large enough to cover
|
||
the time spent for sending, unexpectedly extended RTT
|
||
time, and the time needed to retransmit the lost UDP
|
||
packet. The effective latency value will be the maximum
|
||
of this options’ value and the value of <samp>peerlatency</samp>
|
||
set by the peer side. Before version 1.3.0 this option
|
||
is only available as <samp>latency</samp>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>recv_buffer_size=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set UDP receive buffer size, expressed in bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>send_buffer_size=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set UDP send buffer size, expressed in bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>timeout=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>Set raise error timeouts for read, write and connect operations. Note that the
|
||
SRT library has internal timeouts which can be controlled separately, the
|
||
value set here is only a cap on those.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tlpktdrop=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Too-late Packet Drop. When enabled on receiver, it skips
|
||
missing packets that have not been delivered in time and
|
||
delivers the following packets to the application when
|
||
their time-to-play has come. It also sends a fake ACK to
|
||
the sender. When enabled on sender and enabled on the
|
||
receiving peer, the sender drops the older packets that
|
||
have no chance of being delivered in time. It was
|
||
automatically enabled in the sender if the receiver
|
||
supports it.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>sndbuf=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set send buffer size, expressed in bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>rcvbuf=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set receive buffer size, expressed in bytes.
|
||
</p>
|
||
<p>Receive buffer must not be greater than <samp>ffs</samp>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>lossmaxttl=<var>packets</var></samp></span></dt>
|
||
<dd><p>The value up to which the Reorder Tolerance may grow. When
|
||
Reorder Tolerance is > 0, then packet loss report is delayed
|
||
until that number of packets come in. Reorder Tolerance
|
||
increases every time a "belated" packet has come, but it
|
||
wasn’t due to retransmission (that is, when UDP packets tend
|
||
to come out of order), with the difference between the latest
|
||
sequence and this packet’s sequence, and not more than the
|
||
value of this option. By default it’s 0, which means that this
|
||
mechanism is turned off, and the loss report is always sent
|
||
immediately upon experiencing a "gap" in sequences.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>minversion</samp></span></dt>
|
||
<dd><p>The minimum SRT version that is required from the peer. A connection
|
||
to a peer that does not satisfy the minimum version requirement
|
||
will be rejected.
|
||
</p>
|
||
<p>The version format in hex is 0xXXYYZZ for x.y.z in human readable
|
||
form.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>streamid=<var>string</var></samp></span></dt>
|
||
<dd><p>A string limited to 512 characters that can be set on the socket prior
|
||
to connecting. This stream ID will be able to be retrieved by the
|
||
listener side from the socket that is returned from srt_accept and
|
||
was connected by a socket with that set stream ID. SRT does not enforce
|
||
any special interpretation of the contents of this string.
|
||
This option doesn’t make sense in Rendezvous connection; the result
|
||
might be that simply one side will override the value from the other
|
||
side and it’s the matter of luck which one would win
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>srt_streamid=<var>string</var></samp></span></dt>
|
||
<dd><p>Alias for ‘<samp>streamid</samp>’ to avoid conflict with ffmpeg command line option.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>smoother=<var>live|file</var></samp></span></dt>
|
||
<dd><p>The type of Smoother used for the transmission for that socket, which
|
||
is responsible for the transmission and congestion control. The Smoother
|
||
type must be exactly the same on both connecting parties, otherwise
|
||
the connection is rejected.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>messageapi=<var>1|0</var></samp></span></dt>
|
||
<dd><p>When set, this socket uses the Message API, otherwise it uses Buffer
|
||
API. Note that in live mode (see <samp>transtype</samp>) there’s only
|
||
message API available. In File mode you can chose to use one of two modes:
|
||
</p>
|
||
<p>Stream API (default, when this option is false). In this mode you may
|
||
send as many data as you wish with one sending instruction, or even use
|
||
dedicated functions that read directly from a file. The internal facility
|
||
will take care of any speed and congestion control. When receiving, you
|
||
can also receive as many data as desired, the data not extracted will be
|
||
waiting for the next call. There is no boundary between data portions in
|
||
the Stream mode.
|
||
</p>
|
||
<p>Message API. In this mode your single sending instruction passes exactly
|
||
one piece of data that has boundaries (a message). Contrary to Live mode,
|
||
this message may span across multiple UDP packets and the only size
|
||
limitation is that it shall fit as a whole in the sending buffer. The
|
||
receiver shall use as large buffer as necessary to receive the message,
|
||
otherwise the message will not be given up. When the message is not
|
||
complete (not all packets received or there was a packet loss) it will
|
||
not be given up.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>transtype=<var>live|file</var></samp></span></dt>
|
||
<dd><p>Sets the transmission type for the socket, in particular, setting this
|
||
option sets multiple other parameters to their default values as required
|
||
for a particular transmission type.
|
||
</p>
|
||
<p>live: Set options as for live transmission. In this mode, you should
|
||
send by one sending instruction only so many data that fit in one UDP packet,
|
||
and limited to the value defined first in <samp>payload_size</samp> (1316 is
|
||
default in this mode). There is no speed control in this mode, only the
|
||
bandwidth control, if configured, in order to not exceed the bandwidth with
|
||
the overhead transmission (retransmitted and control packets).
|
||
</p>
|
||
<p>file: Set options as for non-live transmission. See <samp>messageapi</samp>
|
||
for further explanations
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>linger=<var>seconds</var></samp></span></dt>
|
||
<dd><p>The number of seconds that the socket waits for unsent data when closing.
|
||
Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
|
||
seconds in file mode). The range for this option is integers in the
|
||
0 - <code>INT_MAX</code>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tsbpd=<var>1|0</var></samp></span></dt>
|
||
<dd><p>When true, use Timestamp-based Packet Delivery mode. The default behavior
|
||
depends on the transmission type: enabled in live mode, disabled in file
|
||
mode.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>For more information see: <a href="https://github.com/Haivision/srt">https://github.com/Haivision/srt</a>.
|
||
</p>
|
||
<a name="srtp"></a>
|
||
<h3 class="section">3.38 srtp<span class="pull-right"><a class="anchor hidden-xs" href="#srtp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-srtp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Secure Real-time Transport Protocol.
|
||
</p>
|
||
<p>The accepted options are:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>srtp_in_suite</samp></span></dt>
|
||
<dt><span><samp>srtp_out_suite</samp></span></dt>
|
||
<dd><p>Select input and output encoding suites.
|
||
</p>
|
||
<p>Supported values:
|
||
</p><dl compact="compact">
|
||
<dt><span>‘<samp>AES_CM_128_HMAC_SHA1_80</samp>’</span></dt>
|
||
<dt><span>‘<samp>SRTP_AES128_CM_HMAC_SHA1_80</samp>’</span></dt>
|
||
<dt><span>‘<samp>AES_CM_128_HMAC_SHA1_32</samp>’</span></dt>
|
||
<dt><span>‘<samp>SRTP_AES128_CM_HMAC_SHA1_32</samp>’</span></dt>
|
||
</dl>
|
||
|
||
</dd>
|
||
<dt><span><samp>srtp_in_params</samp></span></dt>
|
||
<dt><span><samp>srtp_out_params</samp></span></dt>
|
||
<dd><p>Set input and output encoding parameters, which are expressed by a
|
||
base64-encoded representation of a binary block. The first 16 bytes of
|
||
this binary block are used as master key, the following 14 bytes are
|
||
used as master salt.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="subfile"></a>
|
||
<h3 class="section">3.39 subfile<span class="pull-right"><a class="anchor hidden-xs" href="#subfile" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-subfile" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Virtually extract a segment of a file or another stream.
|
||
The underlying stream must be seekable.
|
||
</p>
|
||
<p>Accepted options:
|
||
</p><dl compact="compact">
|
||
<dt><span><samp>start</samp></span></dt>
|
||
<dd><p>Start offset of the extracted segment, in bytes.
|
||
</p></dd>
|
||
<dt><span><samp>end</samp></span></dt>
|
||
<dd><p>End offset of the extracted segment, in bytes.
|
||
If set to 0, extract till end of file.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>Examples:
|
||
</p>
|
||
<p>Extract a chapter from a DVD VOB file (start and end sectors obtained
|
||
externally and multiplied by 2048):
|
||
</p><div class="example">
|
||
<pre class="example">subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
|
||
</pre></div>
|
||
|
||
<p>Play an AVI file directly from a TAR archive:
|
||
</p><div class="example">
|
||
<pre class="example">subfile,,start,183241728,end,366490624,,:archive.tar
|
||
</pre></div>
|
||
|
||
<p>Play a MPEG-TS file from start offset till end:
|
||
</p><div class="example">
|
||
<pre class="example">subfile,,start,32815239,end,0,,:video.ts
|
||
</pre></div>
|
||
|
||
<a name="tee"></a>
|
||
<h3 class="section">3.40 tee<span class="pull-right"><a class="anchor hidden-xs" href="#tee" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-tee" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Writes the output to multiple protocols. The individual outputs are separated
|
||
by |
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">tee:file://path/to/local/this.avi|file://path/to/local/that.avi
|
||
</pre></div>
|
||
|
||
<a name="tcp"></a>
|
||
<h3 class="section">3.41 tcp<span class="pull-right"><a class="anchor hidden-xs" href="#tcp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-tcp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Transmission Control Protocol.
|
||
</p>
|
||
<p>The required syntax for a TCP url is:
|
||
</p><div class="example">
|
||
<pre class="example">tcp://<var>hostname</var>:<var>port</var>[?<var>options</var>]
|
||
</pre></div>
|
||
|
||
<p><var>options</var> contains a list of &-separated options of the form
|
||
<var>key</var>=<var>val</var>.
|
||
</p>
|
||
<p>The list of supported options follows.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>listen=<var>2|1|0</var></samp></span></dt>
|
||
<dd><p>Listen for an incoming connection. 0 disables listen, 1 enables listen in
|
||
single client mode, 2 enables listen in multi-client mode. Default value is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>local_addr=<var>addr</var></samp></span></dt>
|
||
<dd><p>Local IP address of a network interface used for tcp socket connect.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>local_port=<var>port</var></samp></span></dt>
|
||
<dd><p>Local port used for tcp socket connect.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>timeout=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>Set raise error timeout, expressed in microseconds.
|
||
</p>
|
||
<p>This option is only relevant in read mode: if no data arrived in more
|
||
than this time interval, raise error.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>listen_timeout=<var>milliseconds</var></samp></span></dt>
|
||
<dd><p>Set listen timeout, expressed in milliseconds.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>recv_buffer_size=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set receive buffer size, expressed bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>send_buffer_size=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set send buffer size, expressed bytes.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tcp_nodelay=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0.
|
||
</p>
|
||
<p><em>Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.</em>
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tcp_mss=<var>bytes</var></samp></span></dt>
|
||
<dd><p>Set maximum segment size for outgoing TCP packets, expressed in bytes.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<p>The following example shows how to setup a listening TCP connection
|
||
with <code>ffmpeg</code>, which is then accessed with <code>ffplay</code>:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -i <var>input</var> -f <var>format</var> tcp://<var>hostname</var>:<var>port</var>?listen
|
||
ffplay tcp://<var>hostname</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
<a name="tls"></a>
|
||
<h3 class="section">3.42 tls<span class="pull-right"><a class="anchor hidden-xs" href="#tls" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-tls" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
|
||
</p>
|
||
<p>The required syntax for a TLS/SSL url is:
|
||
</p><div class="example">
|
||
<pre class="example">tls://<var>hostname</var>:<var>port</var>[?<var>options</var>]
|
||
</pre></div>
|
||
|
||
<p>The following parameters can be set via command line options
|
||
(or in code via <code>AVOption</code>s):
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>ca_file, cafile=<var>filename</var></samp></span></dt>
|
||
<dd><p>A file containing certificate authority (CA) root certificates to treat
|
||
as trusted. If the linked TLS library contains a default this might not
|
||
need to be specified for verification to work, but not all libraries and
|
||
setups have defaults built in.
|
||
The file must be in OpenSSL PEM format.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>tls_verify=<var>1|0</var></samp></span></dt>
|
||
<dd><p>If enabled, try to verify the peer that we are communicating with.
|
||
Note, if using OpenSSL, this currently only makes sure that the
|
||
peer certificate is signed by one of the root certificates in the CA
|
||
database, but it does not validate that the certificate actually
|
||
matches the host name we are trying to connect to. (With other backends,
|
||
the host name is validated as well.)
|
||
</p>
|
||
<p>This is disabled by default since it requires a CA database to be
|
||
provided by the caller in many cases.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>cert_file, cert=<var>filename</var></samp></span></dt>
|
||
<dd><p>A file containing a certificate to use in the handshake with the peer.
|
||
(When operating as server, in listen mode, this is more often required
|
||
by the peer, while client certificates only are mandated in certain
|
||
setups.)
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>key_file, key=<var>filename</var></samp></span></dt>
|
||
<dd><p>A file containing the private key for the certificate.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>listen=<var>1|0</var></samp></span></dt>
|
||
<dd><p>If enabled, listen for connections on the provided port, and assume
|
||
the server role in the handshake instead of the client role.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>http_proxy</samp></span></dt>
|
||
<dd><p>The HTTP proxy to tunnel through, e.g. <code>http://example.com:1234</code>.
|
||
The proxy must support the CONNECT method.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
<p>Example command lines:
|
||
</p>
|
||
<p>To create a TLS/SSL server that serves an input stream.
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -i <var>input</var> -f <var>format</var> tls://<var>hostname</var>:<var>port</var>?listen&cert=<var>server.crt</var>&key=<var>server.key</var>
|
||
</pre></div>
|
||
|
||
<p>To play back a stream from the TLS/SSL server using <code>ffplay</code>:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">ffplay tls://<var>hostname</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
<a name="udp"></a>
|
||
<h3 class="section">3.43 udp<span class="pull-right"><a class="anchor hidden-xs" href="#udp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-udp" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>User Datagram Protocol.
|
||
</p>
|
||
<p>The required syntax for an UDP URL is:
|
||
</p><div class="example">
|
||
<pre class="example">udp://<var>hostname</var>:<var>port</var>[?<var>options</var>]
|
||
</pre></div>
|
||
|
||
<p><var>options</var> contains a list of &-separated options of the form <var>key</var>=<var>val</var>.
|
||
</p>
|
||
<p>In case threading is enabled on the system, a circular buffer is used
|
||
to store the incoming data, which allows one to reduce loss of data due to
|
||
UDP socket buffer overruns. The <var>fifo_size</var> and
|
||
<var>overrun_nonfatal</var> options are related to this buffer.
|
||
</p>
|
||
<p>The list of supported options follows.
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>buffer_size=<var>size</var></samp></span></dt>
|
||
<dd><p>Set the UDP maximum socket buffer size in bytes. This is used to set either
|
||
the receive or send buffer size, depending on what the socket is used for.
|
||
Default is 32 KB for output, 384 KB for input. See also <var>fifo_size</var>.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>bitrate=<var>bitrate</var></samp></span></dt>
|
||
<dd><p>If set to nonzero, the output will have the specified constant bitrate if the
|
||
input has enough packets to sustain it.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>burst_bits=<var>bits</var></samp></span></dt>
|
||
<dd><p>When using <var>bitrate</var> this specifies the maximum number of bits in
|
||
packet bursts.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>localport=<var>port</var></samp></span></dt>
|
||
<dd><p>Override the local UDP port to bind with.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>localaddr=<var>addr</var></samp></span></dt>
|
||
<dd><p>Local IP address of a network interface used for sending packets or joining
|
||
multicast groups.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>pkt_size=<var>size</var></samp></span></dt>
|
||
<dd><p>Set the size in bytes of UDP packets.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>reuse=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Explicitly allow or disallow reusing UDP sockets.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>ttl=<var>ttl</var></samp></span></dt>
|
||
<dd><p>Set the time to live value (for multicast only).
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>connect=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Initialize the UDP socket with <code>connect()</code>. In this case, the
|
||
destination address can’t be changed with ff_udp_set_remote_url later.
|
||
If the destination address isn’t known at the start, this option can
|
||
be specified in ff_udp_set_remote_url, too.
|
||
This allows finding out the source address for the packets with getsockname,
|
||
and makes writes return with AVERROR(ECONNREFUSED) if "destination
|
||
unreachable" is received.
|
||
For receiving, this gives the benefit of only receiving packets from
|
||
the specified peer address/port.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>sources=<var>address</var>[,<var>address</var>]</samp></span></dt>
|
||
<dd><p>Only receive packets sent from the specified addresses. In case of multicast,
|
||
also subscribe to multicast traffic coming from these addresses only.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>block=<var>address</var>[,<var>address</var>]</samp></span></dt>
|
||
<dd><p>Ignore packets sent from the specified addresses. In case of multicast, also
|
||
exclude the source addresses in the multicast subscription.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>fifo_size=<var>units</var></samp></span></dt>
|
||
<dd><p>Set the UDP receiving circular buffer size, expressed as a number of
|
||
packets with size of 188 bytes. If not specified defaults to 7*4096.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>overrun_nonfatal=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Survive in case of UDP receiving circular buffer overrun. Default
|
||
value is 0.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>timeout=<var>microseconds</var></samp></span></dt>
|
||
<dd><p>Set raise error timeout, expressed in microseconds.
|
||
</p>
|
||
<p>This option is only relevant in read mode: if no data arrived in more
|
||
than this time interval, raise error.
|
||
</p>
|
||
</dd>
|
||
<dt><span><samp>broadcast=<var>1|0</var></samp></span></dt>
|
||
<dd><p>Explicitly allow or disallow UDP broadcasting.
|
||
</p>
|
||
<p>Note that broadcasting may not work properly on networks having
|
||
a broadcast storm protection.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="Examples-1"></a>
|
||
<h4 class="subsection">3.43.1 Examples<span class="pull-right"><a class="anchor hidden-xs" href="#Examples-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Examples-1" aria-hidden="true">TOC</a></span></h4>
|
||
|
||
<ul>
|
||
<li> Use <code>ffmpeg</code> to stream over UDP to a remote endpoint:
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -i <var>input</var> -f <var>format</var> udp://<var>hostname</var>:<var>port</var>
|
||
</pre></div>
|
||
|
||
</li><li> Use <code>ffmpeg</code> to stream in mpegts format over UDP using 188
|
||
sized UDP packets, using a large input buffer:
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -i <var>input</var> -f mpegts udp://<var>hostname</var>:<var>port</var>?pkt_size=188&buffer_size=65535
|
||
</pre></div>
|
||
|
||
</li><li> Use <code>ffmpeg</code> to receive over UDP from a remote endpoint:
|
||
<div class="example">
|
||
<pre class="example">ffmpeg -i udp://[<var>multicast-address</var>]:<var>port</var> ...
|
||
</pre></div>
|
||
</li></ul>
|
||
|
||
<a name="unix"></a>
|
||
<h3 class="section">3.44 unix<span class="pull-right"><a class="anchor hidden-xs" href="#unix" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-unix" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>Unix local socket
|
||
</p>
|
||
<p>The required syntax for a Unix socket URL is:
|
||
</p>
|
||
<div class="example">
|
||
<pre class="example">unix://<var>filepath</var>
|
||
</pre></div>
|
||
|
||
<p>The following parameters can be set via command line options
|
||
(or in code via <code>AVOption</code>s):
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>timeout</samp></span></dt>
|
||
<dd><p>Timeout in ms.
|
||
</p></dd>
|
||
<dt><span><samp>listen</samp></span></dt>
|
||
<dd><p>Create the Unix socket in listening mode.
|
||
</p></dd>
|
||
</dl>
|
||
|
||
<a name="zmq"></a>
|
||
<h3 class="section">3.45 zmq<span class="pull-right"><a class="anchor hidden-xs" href="#zmq" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-zmq" aria-hidden="true">TOC</a></span></h3>
|
||
|
||
<p>ZeroMQ asynchronous messaging using the libzmq library.
|
||
</p>
|
||
<p>This library supports unicast streaming to multiple clients without relying on
|
||
an external server.
|
||
</p>
|
||
<p>The required syntax for streaming or connecting to a stream is:
|
||
</p><div class="example">
|
||
<pre class="example">zmq:tcp://ip-address:port
|
||
</pre></div>
|
||
|
||
<p>Example:
|
||
Create a localhost stream on port 5555:
|
||
</p><div class="example">
|
||
<pre class="example">ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
|
||
</pre></div>
|
||
|
||
<p>Multiple clients may connect to the stream using:
|
||
</p><div class="example">
|
||
<pre class="example">ffplay zmq:tcp://127.0.0.1:5555
|
||
</pre></div>
|
||
|
||
<p>Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
|
||
The server side binds to a port and publishes data. Clients connect to the
|
||
server (via IP address/port) and subscribe to the stream. The order in which
|
||
the server and client start generally does not matter.
|
||
</p>
|
||
<p>ffmpeg must be compiled with the –enable-libzmq option to support
|
||
this protocol.
|
||
</p>
|
||
<p>Options can be set on the <code>ffmpeg</code>/<code>ffplay</code> command
|
||
line. The following options are supported:
|
||
</p>
|
||
<dl compact="compact">
|
||
<dt><span><samp>pkt_size</samp></span></dt>
|
||
<dd><p>Forces the maximum packet size for sending/receiving data. The default value is
|
||
131,072 bytes. On the server side, this sets the maximum size of sent packets
|
||
via ZeroMQ. On the clients, it sets an internal buffer size for receiving
|
||
packets. Note that pkt_size on the clients should be equal to or greater than
|
||
pkt_size on the server. Otherwise the received message may be truncated causing
|
||
decoding errors.
|
||
</p>
|
||
</dd>
|
||
</dl>
|
||
|
||
|
||
<a name="See-Also"></a>
|
||
<h2 class="chapter">4 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="libavformat.html">libavformat</a>
|
||
</p>
|
||
|
||
<a name="Authors"></a>
|
||
<h2 class="chapter">5 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>
|