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diff --git a/ffmpeg/doc/ffmpeg-protocols.html b/ffmpeg/doc/ffmpeg-protocols.html new file mode 100644 index 0000000..99602bf --- /dev/null +++ b/ffmpeg/doc/ffmpeg-protocols.html @@ -0,0 +1,2521 @@ +<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> +<html> +<!-- Created by GNU Texinfo 6.8, https://www.gnu.org/software/texinfo/ --> + <head> + <meta charset="utf-8"> + <title> + FFmpeg Protocols Documentation + </title> + <meta name="viewport" content="width=device-width,initial-scale=1.0"> + <link rel="stylesheet" type="text/css" href="bootstrap.min.css"> + <link rel="stylesheet" type="text/css" href="style.min.css"> + </head> + <body> + <div class="container"> + <h1> + FFmpeg Protocols Documentation + </h1> +<div align="center"> +</div> + + +<a name="SEC_Top"></a> + +<div class="Contents_element" id="SEC_Contents"> +<h2 class="contents-heading">Table of Contents</h2> + +<div class="contents"> + +<ul class="no-bullet"> + <li><a id="toc-Description" href="#Description">1 Description</a></li> + <li><a id="toc-Protocol-Options" href="#Protocol-Options">2 Protocol Options</a></li> + <li><a id="toc-Protocols" href="#Protocols">3 Protocols</a> + <ul class="no-bullet"> + <li><a id="toc-amqp" href="#amqp">3.1 amqp</a></li> + <li><a id="toc-async" href="#async">3.2 async</a></li> + <li><a id="toc-bluray" href="#bluray">3.3 bluray</a></li> + <li><a id="toc-cache" href="#cache">3.4 cache</a></li> + <li><a id="toc-concat" href="#concat">3.5 concat</a></li> + <li><a id="toc-concatf" href="#concatf">3.6 concatf</a></li> + <li><a id="toc-crypto" href="#crypto">3.7 crypto</a></li> + <li><a id="toc-data" href="#data">3.8 data</a></li> + <li><a id="toc-file" href="#file">3.9 file</a></li> + <li><a id="toc-ftp" href="#ftp">3.10 ftp</a></li> + <li><a id="toc-gopher" href="#gopher">3.11 gopher</a></li> + <li><a id="toc-gophers" href="#gophers">3.12 gophers</a></li> + <li><a id="toc-hls" href="#hls">3.13 hls</a></li> + <li><a id="toc-http" href="#http">3.14 http</a> + <ul class="no-bullet"> + <li><a id="toc-HTTP-Cookies" href="#HTTP-Cookies">3.14.1 HTTP Cookies</a></li> + </ul></li> + <li><a id="toc-Icecast" href="#Icecast">3.15 Icecast</a></li> + <li><a id="toc-ipfs" href="#ipfs">3.16 ipfs</a></li> + <li><a id="toc-mmst" href="#mmst">3.17 mmst</a></li> + <li><a id="toc-mmsh" href="#mmsh">3.18 mmsh</a></li> + <li><a id="toc-md5" href="#md5">3.19 md5</a></li> + <li><a id="toc-pipe" href="#pipe">3.20 pipe</a></li> + <li><a id="toc-prompeg" href="#prompeg">3.21 prompeg</a></li> + <li><a id="toc-rist" href="#rist">3.22 rist</a></li> + <li><a id="toc-rtmp" href="#rtmp">3.23 rtmp</a></li> + <li><a id="toc-rtmpe" href="#rtmpe">3.24 rtmpe</a></li> + <li><a id="toc-rtmps" href="#rtmps">3.25 rtmps</a></li> + <li><a id="toc-rtmpt" href="#rtmpt">3.26 rtmpt</a></li> + <li><a id="toc-rtmpte" href="#rtmpte">3.27 rtmpte</a></li> + <li><a id="toc-rtmpts" href="#rtmpts">3.28 rtmpts</a></li> + <li><a id="toc-libsmbclient" href="#libsmbclient">3.29 libsmbclient</a></li> + <li><a id="toc-libssh" href="#libssh">3.30 libssh</a></li> + <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">3.31 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte</a></li> + <li><a id="toc-rtp" href="#rtp">3.32 rtp</a></li> + <li><a id="toc-rtsp" href="#rtsp">3.33 rtsp</a> + <ul class="no-bullet"> + <li><a id="toc-Muxer" href="#Muxer">3.33.1 Muxer</a></li> + <li><a id="toc-Demuxer" href="#Demuxer">3.33.2 Demuxer</a></li> + <li><a id="toc-Examples" href="#Examples">3.33.3 Examples</a></li> + </ul></li> + <li><a id="toc-sap" href="#sap">3.34 sap</a> + <ul class="no-bullet"> + <li><a id="toc-Muxer-1" href="#Muxer-1">3.34.1 Muxer</a></li> + <li><a id="toc-Demuxer-1" href="#Demuxer-1">3.34.2 Demuxer</a></li> + </ul></li> + <li><a id="toc-sctp" href="#sctp">3.35 sctp</a></li> + <li><a id="toc-srt" href="#srt">3.36 srt</a></li> + <li><a id="toc-srtp" href="#srtp">3.37 srtp</a></li> + <li><a id="toc-subfile" href="#subfile">3.38 subfile</a></li> + <li><a id="toc-tee" href="#tee">3.39 tee</a></li> + <li><a id="toc-tcp" href="#tcp">3.40 tcp</a></li> + <li><a id="toc-tls" href="#tls">3.41 tls</a></li> + <li><a id="toc-udp" href="#udp">3.42 udp</a> + <ul class="no-bullet"> + <li><a id="toc-Examples-1" href="#Examples-1">3.42.1 Examples</a></li> + </ul></li> + <li><a id="toc-unix" href="#unix">3.43 unix</a></li> + <li><a id="toc-zmq" href="#zmq">3.44 zmq</a></li> + </ul></li> + <li><a id="toc-See-Also" href="#See-Also">4 See Also</a></li> + <li><a id="toc-Authors" href="#Authors">5 Authors</a></li> +</ul> +</div> +</div> + +<a name="Description"></a> +<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2> + +<p>This document describes the input and output protocols provided by the +libavformat library. +</p> + +<a name="Protocol-Options"></a> +<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> + +<p>The libavformat library provides some generic global options, which +can be set on all the protocols. In addition each protocol may support +so-called private options, which are specific for that component. +</p> +<p>Options may be set by specifying -<var>option</var> <var>value</var> in the +FFmpeg tools, or by setting the value explicitly in the +<code>AVFormatContext</code> options or using the <samp>libavutil/opt.h</samp> API +for programmatic use. +</p> +<p>The list of supported options follows: +</p> +<dl compact="compact"> +<dt><span><samp>protocol_whitelist <var>list</var> (<em>input</em>)</samp></span></dt> +<dd><p>Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols +prefixed by "-" are disabled. +All protocols are allowed by default but protocols used by an another +protocol (nested protocols) are restricted to a per protocol subset. +</p></dd> +</dl> + + +<a name="Protocols"></a> +<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> + +<p>Protocols are configured elements in FFmpeg that enable access to +resources that require specific protocols. +</p> +<p>When you configure your FFmpeg build, all the supported protocols are +enabled by default. You can list all available ones using the +configure option "–list-protocols". +</p> +<p>You can disable all the protocols using the configure option +"–disable-protocols", and selectively enable a protocol using the +option "–enable-protocol=<var>PROTOCOL</var>", or you can disable a +particular protocol using the option +"–disable-protocol=<var>PROTOCOL</var>". +</p> +<p>The option "-protocols" of the ff* tools will display the list of +supported protocols. +</p> +<p>All protocols accept the following options: +</p> +<dl compact="compact"> +<dt><span><samp>rw_timeout</samp></span></dt> +<dd><p>Maximum time to wait for (network) read/write operations to complete, +in microseconds. +</p></dd> +</dl> + +<p>A description of the currently available protocols follows. +</p> +<a name="amqp"></a> +<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> + +<p>Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based +publish-subscribe communication protocol. +</p> +<p>FFmpeg must be compiled with –enable-librabbitmq to support AMQP. A separate +AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ. +</p> +<p>After starting the broker, an FFmpeg client may stream data to the broker using +the command: +</p> +<div class="example"> +<pre class="example">ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost] +</pre></div> + +<p>Where hostname and port (default is 5672) is the address of the broker. The +client may also set a user/password for authentication. The default for both +fields is "guest". Name of virtual host on broker can be set with vhost. The +default value is "/". +</p> +<p>Muliple subscribers may stream from the broker using the command: +</p><div class="example"> +<pre class="example">ffplay amqp://[[user]:[password]@]hostname[:port][/vhost] +</pre></div> + +<p>In RabbitMQ all data published to the broker flows through a specific exchange, +and each subscribing client has an assigned queue/buffer. When a packet arrives +at an exchange, it may be copied to a client’s queue depending on the exchange +and routing_key fields. +</p> +<p>The following options are supported: +</p> +<dl compact="compact"> +<dt><span><samp>exchange</samp></span></dt> +<dd><p>Sets the exchange to use on the broker. RabbitMQ has several predefined +exchanges: "amq.direct" is the default exchange, where the publisher and +subscriber must have a matching routing_key; "amq.fanout" is the same as a +broadcast operation (i.e. the data is forwarded to all queues on the fanout +exchange independent of the routing_key); and "amq.topic" is similar to +"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ +documentation). +</p> +</dd> +<dt><span><samp>routing_key</samp></span></dt> +<dd><p>Sets the routing key. The default value is "amqp". The routing key is used on +the "amq.direct" and "amq.topic" exchanges to decide whether packets are written +to the queue of a subscriber. +</p> +</dd> +<dt><span><samp>pkt_size</samp></span></dt> +<dd><p>Maximum size of each packet sent/received to the broker. Default is 131072. +Minimum is 4096 and max is any large value (representable by an int). When +receiving packets, this sets an internal buffer size in FFmpeg. It should be +equal to or greater than the size of the published packets to the broker. Otherwise +the received message may be truncated causing decoding errors. +</p> +</dd> +<dt><span><samp>connection_timeout</samp></span></dt> +<dd><p>The timeout in seconds during the initial connection to the broker. The +default value is rw_timeout, or 5 seconds if rw_timeout is not set. +</p> +</dd> +<dt><span><samp>delivery_mode <var>mode</var></samp></span></dt> +<dd><p>Sets the delivery mode of each message sent to broker. +The following values are accepted: +</p><dl compact="compact"> +<dt><span>‘<samp>persistent</samp>’</span></dt> +<dd><p>Delivery mode set to "persistent" (2). This is the default value. +Messages may be written to the broker’s disk depending on its setup. +</p> +</dd> +<dt><span>‘<samp>non-persistent</samp>’</span></dt> +<dd><p>Delivery mode set to "non-persistent" (1). +Messages will stay in broker’s memory unless the broker is under memory +pressure. +</p> +</dd> +</dl> + +</dd> +</dl> + +<a name="async"></a> +<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> + +<p>Asynchronous data filling wrapper for input stream. +</p> +<p>Fill data in a background thread, to decouple I/O operation from demux thread. +</p> +<div class="example"> +<pre class="example">async:<var>URL</var> +async:http://host/resource +async:cache:http://host/resource +</pre></div> + +<a name="bluray"></a> +<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> + +<p>Read BluRay playlist. +</p> +<p>The accepted options are: +</p><dl compact="compact"> +<dt><span><samp>angle</samp></span></dt> +<dd><p>BluRay angle +</p> +</dd> +<dt><span><samp>chapter</samp></span></dt> +<dd><p>Start chapter (1...N) +</p> +</dd> +<dt><span><samp>playlist</samp></span></dt> +<dd><p>Playlist to read (BDMV/PLAYLIST/?????.mpls) +</p> +</dd> +</dl> + +<p>Examples: +</p> +<p>Read longest playlist from BluRay mounted to /mnt/bluray: +</p><div class="example"> +<pre class="example">bluray:/mnt/bluray +</pre></div> + +<p>Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: +</p><div class="example"> +<pre class="example">-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray +</pre></div> + +<a name="cache"></a> +<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> + +<p>Caching wrapper for input stream. +</p> +<p>Cache the input stream to temporary file. It brings seeking capability to live streams. +</p> +<p>The accepted options are: +</p><dl compact="compact"> +<dt><span><samp>read_ahead_limit</samp></span></dt> +<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="file"></a> +<h3 class="section">3.9 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">3.10 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">3.11 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">3.12 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">3.13 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">3.14 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">3.14.1 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">3.15 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">3.16 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>If a user doesn’t have a node of their own then the public gateway <code>https://dweb.link</code> +is used by default. +</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. If that fails <code>https://dweb.link</code> will be used. +</p> +</dd> +</dl> + +<p>One can use this protocol in 2 ways. Using IPFS: +</p><div class="example"> +<pre class="example">ffplay ipfs://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T +</pre></div> + +<p>Or the IPNS protocol (IPNS is mutable IPFS): +</p><div class="example"> +<pre class="example">ffplay ipns://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T +</pre></div> + +<a name="mmst"></a> +<h3 class="section">3.17 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">3.18 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">3.19 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">3.20 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><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> +</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">3.21 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">3.22 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">3.23 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_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">3.24 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">3.25 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">3.26 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">3.27 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">3.28 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">3.29 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">3.30 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">3.31 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">3.32 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">3.33 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">3.33.1 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">3.33.2 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">3.33.3 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">3.34 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">3.34.1 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">3.34.2 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">3.35 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">3.36 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">3.37 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">3.38 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">3.39 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">3.40 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>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">3.41 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">3.42 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">3.42.1 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">3.43 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">3.44 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 +(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command +<code>git log</code> in the FFmpeg source directory, or browsing the +online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>. +</p> +<p>Maintainers for the specific components are listed in the file +<samp>MAINTAINERS</samp> in the source code tree. +</p> + + <p style="font-size: small;"> + This document was generated using <a href="https://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>. + </p> + </div> + </body> +</html> |