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| author | Xavier Del Campo <xavi.dcr@gmail.com> | 2017-02-04 14:49:08 +0100 |
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| committer | Xavier Del Campo <xavi.dcr@gmail.com> | 2017-02-04 14:49:08 +0100 |
| commit | 189ecf754d0c8131464bfdff98fb56e7752556b1 (patch) | |
| tree | 89e7d02128bbc7b2d3f5c19a3da14ec14291982a /Music/ffmpeg/doc/ffmpeg-protocols.html | |
| download | airport-189ecf754d0c8131464bfdff98fb56e7752556b1.tar.gz | |
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diff --git a/Music/ffmpeg/doc/ffmpeg-protocols.html b/Music/ffmpeg/doc/ffmpeg-protocols.html new file mode 100755 index 0000000..fe0adab --- /dev/null +++ b/Music/ffmpeg/doc/ffmpeg-protocols.html @@ -0,0 +1,1691 @@ +<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> +<html> +<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ --> + <head> + <meta charset="utf-8"> + <title> + FFmpeg 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> + +<a name="SEC_Contents"></a> +<h2 class="contents-heading">Table of Contents</h2> + +<div class="contents"> + +<ul class="no-bullet"> + <li><a name="toc-Description" href="#Description">1 Description</a></li> + <li><a name="toc-Protocol-Options" href="#Protocol-Options">2 Protocol Options</a></li> + <li><a name="toc-Protocols" href="#Protocols">3 Protocols</a> + <ul class="no-bullet"> + <li><a name="toc-async" href="#async">3.1 async</a></li> + <li><a name="toc-bluray" href="#bluray">3.2 bluray</a></li> + <li><a name="toc-cache" href="#cache">3.3 cache</a></li> + <li><a name="toc-concat" href="#concat">3.4 concat</a></li> + <li><a name="toc-crypto" href="#crypto">3.5 crypto</a></li> + <li><a name="toc-data" href="#data">3.6 data</a></li> + <li><a name="toc-file" href="#file">3.7 file</a></li> + <li><a name="toc-ftp" href="#ftp">3.8 ftp</a></li> + <li><a name="toc-gopher" href="#gopher">3.9 gopher</a></li> + <li><a name="toc-hls" href="#hls">3.10 hls</a></li> + <li><a name="toc-http" href="#http">3.11 http</a> + <ul class="no-bullet"> + <li><a name="toc-HTTP-Cookies" href="#HTTP-Cookies">3.11.1 HTTP Cookies</a></li> + </ul></li> + <li><a name="toc-Icecast" href="#Icecast">3.12 Icecast</a></li> + <li><a name="toc-mmst" href="#mmst">3.13 mmst</a></li> + <li><a name="toc-mmsh" href="#mmsh">3.14 mmsh</a></li> + <li><a name="toc-md5" href="#md5">3.15 md5</a></li> + <li><a name="toc-pipe" href="#pipe">3.16 pipe</a></li> + <li><a name="toc-rtmp" href="#rtmp">3.17 rtmp</a></li> + <li><a name="toc-rtmpe" href="#rtmpe">3.18 rtmpe</a></li> + <li><a name="toc-rtmps" href="#rtmps">3.19 rtmps</a></li> + <li><a name="toc-rtmpt" href="#rtmpt">3.20 rtmpt</a></li> + <li><a name="toc-rtmpte" href="#rtmpte">3.21 rtmpte</a></li> + <li><a name="toc-rtmpts" href="#rtmpts">3.22 rtmpts</a></li> + <li><a name="toc-libsmbclient" href="#libsmbclient">3.23 libsmbclient</a></li> + <li><a name="toc-libssh" href="#libssh">3.24 libssh</a></li> + <li><a name="toc-librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte" href="#librtmp-rtmp_002c-rtmpe_002c-rtmps_002c-rtmpt_002c-rtmpte">3.25 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte</a></li> + <li><a name="toc-rtp" href="#rtp">3.26 rtp</a></li> + <li><a name="toc-rtsp" href="#rtsp">3.27 rtsp</a> + <ul class="no-bullet"> + <li><a name="toc-Examples" href="#Examples">3.27.1 Examples</a></li> + </ul></li> + <li><a name="toc-sap" href="#sap">3.28 sap</a> + <ul class="no-bullet"> + <li><a name="toc-Muxer" href="#Muxer">3.28.1 Muxer</a></li> + <li><a name="toc-Demuxer" href="#Demuxer">3.28.2 Demuxer</a></li> + </ul></li> + <li><a name="toc-sctp" href="#sctp">3.29 sctp</a></li> + <li><a name="toc-srtp" href="#srtp">3.30 srtp</a></li> + <li><a name="toc-subfile" href="#subfile">3.31 subfile</a></li> + <li><a name="toc-tee" href="#tee">3.32 tee</a></li> + <li><a name="toc-tcp" href="#tcp">3.33 tcp</a></li> + <li><a name="toc-tls" href="#tls">3.34 tls</a></li> + <li><a name="toc-udp" href="#udp">3.35 udp</a> + <ul class="no-bullet"> + <li><a name="toc-Examples-1" href="#Examples-1">3.35.1 Examples</a></li> + </ul></li> + <li><a name="toc-unix" href="#unix">3.36 unix</a></li> + </ul></li> + <li><a name="toc-See-Also" href="#See-Also">4 See Also</a></li> + <li><a name="toc-Authors" href="#Authors">5 Authors</a></li> +</ul> +</div> + + +<a name="Description"></a> +<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2> + +<p>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>The list of supported options follows: +</p> +<dl compact="compact"> +<dt><samp>protocol_whitelist <var>list</var> (<em>input</em>)</samp></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><samp>rw_timeout</samp></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="async"></a> +<h3 class="section">3.1 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">3.2 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><samp>angle</samp></dt> +<dd><p>BluRay angle +</p> +</dd> +<dt><samp>chapter</samp></dt> +<dd><p>Start chapter (1...N) +</p> +</dd> +<dt><samp>playlist</samp></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">3.3 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> +<div class="example"> +<pre class="example">cache:<var>URL</var> +</pre></div> + +<a name="concat"></a> +<h3 class="section">3.4 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="crypto"></a> +<h3 class="section">3.5 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><samp>key</samp></dt> +<dd><p>Set the AES decryption key binary block from given hexadecimal representation. +</p> +</dd> +<dt><samp>iv</samp></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">3.6 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">3.7 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><samp>truncate</samp></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><samp>blocksize</samp></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> +</dl> + +<a name="ftp"></a> +<h3 class="section">3.8 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><samp>timeout</samp></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><samp>ftp-anonymous-password</samp></dt> +<dd><p>Password used when login as anonymous user. Typically an e-mail address +should be used. +</p> +</dd> +<dt><samp>ftp-write-seekable</samp></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> +<p>This protocol accepts the following options: +</p> +<dl compact="compact"> +<dt><samp>follow</samp></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> +</dl> + +<a name="gopher"></a> +<h3 class="section">3.9 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="hls"></a> +<h3 class="section">3.10 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">3.11 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><samp>seekable</samp></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><samp>chunked_post</samp></dt> +<dd><p>If set to 1 use chunked Transfer-Encoding for posts, default is 1. +</p> +</dd> +<dt><samp>content_type</samp></dt> +<dd><p>Set a specific content type for the POST messages or for listen mode. +</p> +</dd> +<dt><samp>http_proxy</samp></dt> +<dd><p>set HTTP proxy to tunnel through e.g. http://example.com:1234 +</p> +</dd> +<dt><samp>headers</samp></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><samp>multiple_requests</samp></dt> +<dd><p>Use persistent connections if set to 1, default is 0. +</p> +</dd> +<dt><samp>post_data</samp></dt> +<dd><p>Set custom HTTP post data. +</p> +</dd> +<dt><samp>user_agent</samp></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><samp>user-agent</samp></dt> +<dd><p>This is a deprecated option, you can use user_agent instead it. +</p> +</dd> +<dt><samp>timeout</samp></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><samp>reconnect_at_eof</samp></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><samp>reconnect_streamed</samp></dt> +<dd><p>If set then even streamed/non seekable streams will be reconnected on errors. +</p> +</dd> +<dt><samp>reconnect_delay_max</samp></dt> +<dd><p>Sets the maximum delay in seconds after which to give up reconnecting +</p> +</dd> +<dt><samp>mime_type</samp></dt> +<dd><p>Export the MIME type. +</p> +</dd> +<dt><samp>icy</samp></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><samp>icy_metadata_headers</samp></dt> +<dd><p>If the server supports ICY metadata, this contains the ICY-specific HTTP reply +headers, separated by newline characters. +</p> +</dd> +<dt><samp>icy_metadata_packet</samp></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><samp>cookies</samp></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><samp>offset</samp></dt> +<dd><p>Set initial byte offset. +</p> +</dd> +<dt><samp>end_offset</samp></dt> +<dd><p>Try to limit the request to bytes preceding this offset. +</p> +</dd> +<dt><samp>method</samp></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><samp>listen</samp></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 or ffserver.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> +</dl> + +<a name="HTTP-Cookies"></a> +<h4 class="subsection">3.11.1 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">3.12 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><samp>ice_genre</samp></dt> +<dd><p>Set the stream genre. +</p> +</dd> +<dt><samp>ice_name</samp></dt> +<dd><p>Set the stream name. +</p> +</dd> +<dt><samp>ice_description</samp></dt> +<dd><p>Set the stream description. +</p> +</dd> +<dt><samp>ice_url</samp></dt> +<dd><p>Set the stream website URL. +</p> +</dd> +<dt><samp>ice_public</samp></dt> +<dd><p>Set if the stream should be public. +The default is 0 (not public). +</p> +</dd> +<dt><samp>user_agent</samp></dt> +<dd><p>Override the User-Agent header. If not specified a string of the form +"Lavf/<version>" will be used. +</p> +</dd> +<dt><samp>password</samp></dt> +<dd><p>Set the Icecast mountpoint password. +</p> +</dd> +<dt><samp>content_type</samp></dt> +<dd><p>Set the stream content type. This must be set if it is different from +audio/mpeg. +</p> +</dd> +<dt><samp>legacy_icecast</samp></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> +</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="mmst"></a> +<h3 class="section">3.13 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">3.14 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">3.15 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">3.16 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><samp>blocksize</samp></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="rtmp"></a> +<h3 class="section">3.17 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><samp>username</samp></dt> +<dd><p>An optional username (mostly for publishing). +</p> +</dd> +<dt><samp>password</samp></dt> +<dd><p>An optional password (mostly for publishing). +</p> +</dd> +<dt><samp>server</samp></dt> +<dd><p>The address of the RTMP server. +</p> +</dd> +<dt><samp>port</samp></dt> +<dd><p>The number of the TCP port to use (by default is 1935). +</p> +</dd> +<dt><samp>app</samp></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><samp>playpath</samp></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><samp>listen</samp></dt> +<dd><p>Act as a server, listening for an incoming connection. +</p> +</dd> +<dt><samp>timeout</samp></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><samp>rtmp_app</samp></dt> +<dd><p>Name of application to connect on the RTMP server. This option +overrides the parameter specified in the URI. +</p> +</dd> +<dt><samp>rtmp_buffer</samp></dt> +<dd><p>Set the client buffer time in milliseconds. The default is 3000. +</p> +</dd> +<dt><samp>rtmp_conn</samp></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><samp>rtmp_flashver</samp></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><samp>rtmp_flush_interval</samp></dt> +<dd><p>Number of packets flushed in the same request (RTMPT only). The default +is 10. +</p> +</dd> +<dt><samp>rtmp_live</samp></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><samp>rtmp_pageurl</samp></dt> +<dd><p>URL of the web page in which the media was embedded. By default no +value will be sent. +</p> +</dd> +<dt><samp>rtmp_playpath</samp></dt> +<dd><p>Stream identifier to play or to publish. This option overrides the +parameter specified in the URI. +</p> +</dd> +<dt><samp>rtmp_subscribe</samp></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><samp>rtmp_swfhash</samp></dt> +<dd><p>SHA256 hash of the decompressed SWF file (32 bytes). +</p> +</dd> +<dt><samp>rtmp_swfsize</samp></dt> +<dd><p>Size of the decompressed SWF file, required for SWFVerification. +</p> +</dd> +<dt><samp>rtmp_swfurl</samp></dt> +<dd><p>URL of the SWF player for the media. By default no value will be sent. +</p> +</dd> +<dt><samp>rtmp_swfverify</samp></dt> +<dd><p>URL to player swf file, compute hash/size automatically. +</p> +</dd> +<dt><samp>rtmp_tcurl</samp></dt> +<dd><p>URL of the target stream. Defaults to proto://host[:port]/app. +</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">3.18 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">3.19 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">3.20 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">3.21 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">3.22 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">3.23 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><samp>timeout</samp></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><samp>truncate</samp></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><samp>workgroup</samp></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">3.24 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><samp>timeout</samp></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><samp>truncate</samp></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><samp>private_key</samp></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">3.25 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">3.26 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><samp>ttl=<var>n</var></samp></dt> +<dd><p>Set the TTL (Time-To-Live) value (for multicast only). +</p> +</dd> +<dt><samp>rtcpport=<var>n</var></samp></dt> +<dd><p>Set the remote RTCP port to <var>n</var>. +</p> +</dd> +<dt><samp>localrtpport=<var>n</var></samp></dt> +<dd><p>Set the local RTP port to <var>n</var>. +</p> +</dd> +<dt><samp>localrtcpport=<var>n</var>'</samp></dt> +<dd><p>Set the local RTCP port to <var>n</var>. +</p> +</dd> +<dt><samp>pkt_size=<var>n</var></samp></dt> +<dd><p>Set max packet size (in bytes) to <var>n</var>. +</p> +</dd> +<dt><samp>connect=0|1</samp></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><samp>sources=<var>ip</var>[,<var>ip</var>]</samp></dt> +<dd><p>List allowed source IP addresses. +</p> +</dd> +<dt><samp>block=<var>ip</var>[,<var>ip</var>]</samp></dt> +<dd><p>List disallowed (blocked) source IP addresses. +</p> +</dd> +<dt><samp>write_to_source=0|1</samp></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><samp>localport=<var>n</var></samp></dt> +<dd><p>Set the local RTP port 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">3.27 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> +<p>The following options are supported. +</p> +<dl compact="compact"> +<dt><samp>initial_pause</samp></dt> +<dd><p>Do not start playing the stream immediately if set to 1. Default value +is 0. +</p> +</dd> +<dt><samp>rtsp_transport</samp></dt> +<dd><p>Set RTSP transport protocols. +</p> +<p>It accepts the following values: +</p><dl compact="compact"> +<dt>‘<samp>udp</samp>’</dt> +<dd><p>Use UDP as lower transport protocol. +</p> +</dd> +<dt>‘<samp>tcp</samp>’</dt> +<dd><p>Use TCP (interleaving within the RTSP control channel) as lower +transport protocol. +</p> +</dd> +<dt>‘<samp>udp_multicast</samp>’</dt> +<dd><p>Use UDP multicast as lower transport protocol. +</p> +</dd> +<dt>‘<samp>http</samp>’</dt> +<dd><p>Use HTTP tunneling as lower transport protocol, which is useful for +passing proxies. +</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><samp>rtsp_flags</samp></dt> +<dd><p>Set RTSP flags. +</p> +<p>The following values are accepted: +</p><dl compact="compact"> +<dt>‘<samp>filter_src</samp>’</dt> +<dd><p>Accept packets only from negotiated peer address and port. +</p></dd> +<dt>‘<samp>listen</samp>’</dt> +<dd><p>Act as a server, listening for an incoming connection. +</p></dd> +<dt>‘<samp>prefer_tcp</samp>’</dt> +<dd><p>Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. +</p></dd> +</dl> + +<p>Default value is ‘<samp>none</samp>’. +</p> +</dd> +<dt><samp>allowed_media_types</samp></dt> +<dd><p>Set media types to accept from the server. +</p> +<p>The following flags are accepted: +</p><dl compact="compact"> +<dt>‘<samp>video</samp>’</dt> +<dt>‘<samp>audio</samp>’</dt> +<dt>‘<samp>data</samp>’</dt> +</dl> + +<p>By default it accepts all media types. +</p> +</dd> +<dt><samp>min_port</samp></dt> +<dd><p>Set minimum local UDP port. Default value is 5000. +</p> +</dd> +<dt><samp>max_port</samp></dt> +<dd><p>Set maximum local UDP port. Default value is 65000. +</p> +</dd> +<dt><samp>timeout</samp></dt> +<dd><p>Set maximum timeout (in seconds) to wait for incoming connections. +</p> +<p>A value of -1 means infinite (default). This option implies the +<samp>rtsp_flags</samp> set to ‘<samp>listen</samp>’. +</p> +</dd> +<dt><samp>reorder_queue_size</samp></dt> +<dd><p>Set number of packets to buffer for handling of reordered packets. +</p> +</dd> +<dt><samp>stimeout</samp></dt> +<dd><p>Set socket TCP I/O timeout in microseconds. +</p> +</dd> +<dt><samp>user-agent</samp></dt> +<dd><p>Override User-Agent header. If not specified, it defaults to the +libavformat identifier string. +</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">3.27.1 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">3.28 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"></a> +<h4 class="subsection">3.28.1 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 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><samp>announce_addr=<var>address</var></samp></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><samp>announce_port=<var>port</var></samp></dt> +<dd><p>Specify the port to send the announcements on, defaults to +9875 if not specified. +</p> +</dd> +<dt><samp>ttl=<var>ttl</var></samp></dt> +<dd><p>Specify the time to live value for the announcements and RTP packets, +defaults to 255. +</p> +</dd> +<dt><samp>same_port=<var>0|1</var></samp></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"></a> +<h4 class="subsection">3.28.2 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 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">3.29 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><samp>listen</samp></dt> +<dd><p>If set to any value, listen for an incoming connection. Outgoing connection is done by default. +</p> +</dd> +<dt><samp>max_streams</samp></dt> +<dd><p>Set the maximum number of streams. By default no limit is set. +</p></dd> +</dl> + +<a name="srtp"></a> +<h3 class="section">3.30 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><samp>srtp_in_suite</samp></dt> +<dt><samp>srtp_out_suite</samp></dt> +<dd><p>Select input and output encoding suites. +</p> +<p>Supported values: +</p><dl compact="compact"> +<dt>‘<samp>AES_CM_128_HMAC_SHA1_80</samp>’</dt> +<dt>‘<samp>SRTP_AES128_CM_HMAC_SHA1_80</samp>’</dt> +<dt>‘<samp>AES_CM_128_HMAC_SHA1_32</samp>’</dt> +<dt>‘<samp>SRTP_AES128_CM_HMAC_SHA1_32</samp>’</dt> +</dl> + +</dd> +<dt><samp>srtp_in_params</samp></dt> +<dt><samp>srtp_out_params</samp></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">3.31 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><samp>start</samp></dt> +<dd><p>Start offset of the extracted segment, in bytes. +</p></dd> +<dt><samp>end</samp></dt> +<dd><p>End offset of the extracted segment, in bytes. +</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> + +<a name="tee"></a> +<h3 class="section">3.32 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">3.33 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><samp>listen=<var>1|0</var></samp></dt> +<dd><p>Listen for an incoming connection. Default value is 0. +</p> +</dd> +<dt><samp>timeout=<var>microseconds</var></samp></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><samp>listen_timeout=<var>milliseconds</var></samp></dt> +<dd><p>Set listen timeout, expressed in milliseconds. +</p> +</dd> +<dt><samp>recv_buffer_size=<var>bytes</var></samp></dt> +<dd><p>Set receive buffer size, expressed bytes. +</p> +</dd> +<dt><samp>send_buffer_size=<var>bytes</var></samp></dt> +<dd><p>Set send buffer size, expressed 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">3.34 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><samp>ca_file, cafile=<var>filename</var></samp></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><samp>tls_verify=<var>1|0</var></samp></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 GnuTLS, +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><samp>cert_file, cert=<var>filename</var></samp></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><samp>key_file, key=<var>filename</var></samp></dt> +<dd><p>A file containing the private key for the certificate. +</p> +</dd> +<dt><samp>listen=<var>1|0</var></samp></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> +</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">3.35 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><samp>buffer_size=<var>size</var></samp></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 64KB. See also <var>fifo_size</var>. +</p> +</dd> +<dt><samp>bitrate=<var>bitrate</var></samp></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><samp>burst_bits=<var>bits</var></samp></dt> +<dd><p>When using <var>bitrate</var> this specifies the maximum number of bits in +packet bursts. +</p> +</dd> +<dt><samp>localport=<var>port</var></samp></dt> +<dd><p>Override the local UDP port to bind with. +</p> +</dd> +<dt><samp>localaddr=<var>addr</var></samp></dt> +<dd><p>Choose the local IP address. This is useful e.g. if sending multicast +and the host has multiple interfaces, where the user can choose +which interface to send on by specifying the IP address of that interface. +</p> +</dd> +<dt><samp>pkt_size=<var>size</var></samp></dt> +<dd><p>Set the size in bytes of UDP packets. +</p> +</dd> +<dt><samp>reuse=<var>1|0</var></samp></dt> +<dd><p>Explicitly allow or disallow reusing UDP sockets. +</p> +</dd> +<dt><samp>ttl=<var>ttl</var></samp></dt> +<dd><p>Set the time to live value (for multicast only). +</p> +</dd> +<dt><samp>connect=<var>1|0</var></samp></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><samp>sources=<var>address</var>[,<var>address</var>]</samp></dt> +<dd><p>Only receive packets sent to the multicast group from one of the +specified sender IP addresses. +</p> +</dd> +<dt><samp>block=<var>address</var>[,<var>address</var>]</samp></dt> +<dd><p>Ignore packets sent to the multicast group from the specified +sender IP addresses. +</p> +</dd> +<dt><samp>fifo_size=<var>units</var></samp></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><samp>overrun_nonfatal=<var>1|0</var></samp></dt> +<dd><p>Survive in case of UDP receiving circular buffer overrun. Default +value is 0. +</p> +</dd> +<dt><samp>timeout=<var>microseconds</var></samp></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><samp>broadcast=<var>1|0</var></samp></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">3.35.1 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">3.36 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><samp>timeout</samp></dt> +<dd><p>Timeout in ms. +</p></dd> +<dt><samp>listen</samp></dt> +<dd><p>Create the Unix socket in listening mode. +</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="ffserver.html">ffserver</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="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>. + </p> + </div> + </body> +</html> |