rtpsessionapi.sgml

ORTP开源的rtp库

<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>user_ts</parameter>&nbsp;:</term>
<listitem><simpara>	a timestamp.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> a rtp packet presented as a mblk_t.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-create-packet" role="function">rtp_session_create_packet ()</title>
<indexterm><primary>rtp_session_create_packet</primary></indexterm><programlisting><link linkend="mblk-t">mblk_t</link>*     rtp_session_create_packet       (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="int">int</link> header_size,
                                             const <link linkend="char">char</link> *payload,
                                             <link linkend="int">int</link> payload_size);</programlisting>
<para>
Allocates a new rtp packet. In the header, ssrc and payload_type according to the session's
context. Timestamp and seq number are not set, there will be set when the packet is going to be
sent with <link linkend="rtp-session-sendm-with-ts"><function>rtp_session_sendm_with_ts()</function></link>.
If payload_size is zero, thus an empty packet (just a RTP header) is returned.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara>		a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>header_size</parameter>&nbsp;:</term>
<listitem><simpara>	the rtp header size. For standart size (without extensions), it is <link linkend="RTP-FIXED-HEADER-SIZE:CAPS"><type>RTP_FIXED_HEADER_SIZE</type></link>
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>payload</parameter>&nbsp;:</term>
<listitem><simpara>data to be copied into the rtp packet.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>payload_size</parameter>&nbsp;:</term>
<listitem><simpara> size of data carried by the rtp packet.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> a rtp packet in a mblk_t (message block) structure.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-create-packet-with-data" role="function">rtp_session_create_packet_with_data ()</title>
<indexterm><primary>rtp_session_create_packet_with_data</primary></indexterm><programlisting><link linkend="mblk-t">mblk_t</link>*     rtp_session_create_packet_with_data
                                            (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="char">char</link> *payload,
                                             <link linkend="int">int</link> payload_size,
                                             <link linkend="void">void</link> (*freefn) (void*));</programlisting>
<para>
Creates a new rtp packet using the given payload buffer (no copy). The header will be allocated separetely.
 In the header, ssrc and payload_type according to the session's
context. Timestamp and seq number are not set, there will be set when the packet is going to be
sent with <link linkend="rtp-session-sendm-with-ts"><function>rtp_session_sendm_with_ts()</function></link>.
oRTP will send this packet using libc's <link linkend="sendmsg"><function>sendmsg()</function></link> (if this function is availlable!) so that there will be no
packet concatenation involving copies to be done in user-space.
 <parameter>freefn</parameter> can be NULL, in that case payload will be kept untouched.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara>		a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>payload</parameter>&nbsp;:</term>
<listitem><simpara> the data to be sent with this packet
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>payload_size</parameter>&nbsp;:</term>
<listitem><simpara> size of data
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>freefn</parameter>&nbsp;:</term>
<listitem><simpara> a function that will be called when the payload buffer is no more needed.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> a rtp packet in a mblk_t (message block) structure.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-sendm-with-ts" role="function">rtp_session_sendm_with_ts ()</title>
<indexterm><primary>rtp_session_sendm_with_ts</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_sendm_with_ts       (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="mblk-t">mblk_t</link> *mp,
                                             <link linkend="uint32-t">uint32_t</link> userts);</programlisting>
<para>
Send the rtp datagram <parameter>mp</parameter> to the destination set by <link linkend="rtp-session-set-remote-addr"><function>rtp_session_set_remote_addr()</function></link> 
with timestamp <parameter>timestamp</parameter>. For audio data, the timestamp is the number
of the first sample resulting of the data transmitted. See rfc1889 for details.
 The packet (<parameter>mp</parameter>) is freed once it is sended.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>mp</parameter>&nbsp;:</term>
<listitem><simpara>	a rtp packet presented as a mblk_t.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>userts</parameter>&nbsp;:</term>
<listitem><simpara>
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> the number of bytes sent over the network.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-get-current-send-ts" role="function">rtp_session_get_current_send_ts ()</title>
<indexterm><primary>rtp_session_get_current_send_ts</primary></indexterm><programlisting><link linkend="uint32-t">uint32_t</link>    rtp_session_get_current_send_ts (<link linkend="RtpSession">RtpSession</link> *session);</programlisting>
<para>
When the rtp session is scheduled and has started to send packets, this function
computes the timestamp that matches to the present time. Using this function can be 
usefull when sending discontinuous streams. Some time can be elapsed between the end
of a stream burst and the begin of a new stream burst, and the application may be not
not aware of this elapsed time. In order to get a valid (current) timestamp to pass to 
#<link linkend="rtp-session-send-with-ts"><function>rtp_session_send_with_ts()</function></link> or #<link linkend="rtp-session-sendm-with-ts"><function>rtp_session_sendm_with_ts()</function></link>, the application may
use <link linkend="rtp-session-get-current-send-ts"><function>rtp_session_get_current_send_ts()</function></link>.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> the current send timestamp for the rtp session.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-flush-sockets" role="function">rtp_session_flush_sockets ()</title>
<indexterm><primary>rtp_session_flush_sockets</primary></indexterm><programlisting><link linkend="void">void</link>        rtp_session_flush_sockets       (<link linkend="RtpSession">RtpSession</link> *session);</programlisting>
<para>
Flushes the sockets for all pending incoming packets.
This can be usefull if you did not listen to the stream for a while
and wishes to start to receive again. During the time no receive is made
packets get bufferised into the internal kernel socket structure.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-set-time-jump-limit" role="function">rtp_session_set_time_jump_limit ()</title>
<indexterm><primary>rtp_session_set_time_jump_limit</primary></indexterm><programlisting><link linkend="void">void</link>        rtp_session_set_time_jump_limit (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="int">int</link> miliseconds);</programlisting>
<para>
oRTP has the possibility to inform the application through a callback registered 
with rtp_session_signal_connect about crazy incoming RTP stream that jumps from 
a timestamp N to N+&lt;some crazy value&gt;. This lets the opportunity for the application
to reset the session in order to resynchronize, or any other action like stopping the call
and reporting an error.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> the rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>miliseconds</parameter>&nbsp;:</term>
<listitem><simpara>


</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-set-recv-buf-size" role="function">rtp_session_set_recv_buf_size ()</title>
<indexterm><primary>rtp_session_set_recv_buf_size</primary></indexterm><programlisting><link linkend="void">void</link>        rtp_session_set_recv_buf_size   (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="int">int</link> bufsize);</programlisting>
<para>
The default value is 65535 bytes, a big value which is working for everyone.
However if your application can make assumption on the MTU, it can be interesting
to set it to a lower value in order to save memory.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>bufsize</parameter>&nbsp;:</term>
<listitem><simpara> buffer size in bytes for receiving packets
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-reset" role="function">rtp_session_reset ()</title>
<indexterm><primary>rtp_session_reset</primary></indexterm><programlisting><link linkend="void">void</link>        rtp_session_reset               (<link linkend="RtpSession">RtpSession</link> *session);</programlisting>
<para>
Reset the session: local and remote addresses are kept unchanged but the internal
queue for ordering and buffering packets is flushed, the session is ready to be
re-synchronised to another incoming stream.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-set-data" role="function">rtp_session_set_data ()</title>
<indexterm><primary>rtp_session_set_data</primary></indexterm><programlisting><link linkend="void">void</link>        rtp_session_set_data            (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="void">void</link> *data);</programlisting>
<para>
Stores some application specific data into the session, so that it is easy to retrieve it
from the signal callbacks using <link linkend="rtp-session-get-data"><function>rtp_session_get_data()</function></link>.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>data</parameter>&nbsp;:</term>
<listitem><simpara> an opaque pointer to be stored in the session
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-get-data" role="function">rtp_session_get_data ()</title>
<indexterm><primary>rtp_session_get_data</primary></indexterm><programlisting><link linkend="void">void</link>*       rtp_session_get_data            (const <link linkend="RtpSession">RtpSession</link> *session);</programlisting>
<para>
</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> the void pointer previously set using <link linkend="rtp-session-set-data"><function>rtp_session_set_data()</function></link>
</simpara></listitem></varlistentry>
</variablelist></refsect2>

</refsect1>




</refentry>