rtpsessionapi.sgml

ORTP开源的rtp库

<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 payload type currently used in outgoing rtp packets
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-get-recv-payload-type" role="function">rtp_session_get_recv_payload_type ()</title>
<indexterm><primary>rtp_session_get_recv_payload_type</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_get_recv_payload_type
                                            (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 payload type currently used in incoming rtp packets
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-set-payload-type" role="function">rtp_session_set_payload_type ()</title>
<indexterm><primary>rtp_session_set_payload_type</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_set_payload_type    (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="int">int</link> pt);</programlisting>
<para>
Sets the expected payload type for incoming packets and payload type to be used for outgoing packets.
If the actual payload type in incoming packets is different that this expected payload type, thus
the "payload_type_changed" signal is emitted.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>pt</parameter>&nbsp;:</term>
<listitem><simpara>
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> 0 on success, -1 if the payload is not defined.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-signal-connect" role="function">rtp_session_signal_connect ()</title>
<indexterm><primary>rtp_session_signal_connect</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_signal_connect      (<link linkend="RtpSession">RtpSession</link> *session,
                                             const <link linkend="char">char</link> *signal,
                                             <link linkend="RtpCallback">RtpCallback</link> cb,
                                             unsigned <link linkend="long-user-data">long user_data</link>);</programlisting>
<para>
This function provides the way for an application to be informed of various events that
may occur during a rtp session. <parameter>signal</parameter> is a string identifying the event, and <parameter>cb</parameter> is 
a user supplied function in charge of processing it. The application can register
several callbacks for the same signal, in the limit of <link linkend="RTP-CALLBACK-TABLE-MAX-ENTRIES:CAPS"><type>RTP_CALLBACK_TABLE_MAX_ENTRIES</type></link>.
Here are name and meaning of supported signals types:
</para>
<para>
"ssrc_changed" : the SSRC of the incoming stream has changed.
</para>
<para>
"payload_type_changed" : the payload type of the incoming stream has changed.
</para>
<para>
"telephone-event_packet" : a telephone-event rtp packet (RFC2833) is received.
</para>
<para>
"telephone-event" : a telephone event has occured. This is a high-level shortcut for "telephone-event_packet".
</para>
<para>
"network_error" : a network error happened on a socket. Arguments of the callback functions are
					a const char * explaining the error, an int errno error code and the user_data as usual.
</para>
<para>
"timestamp_jump" : we have received a packet with timestamp in far future compared to last timestamp received.
					The farness of far future is set by <link linkend="rtp-sesssion-set-time-jump-limit"><function>rtp_sesssion_set_time_jump_limit()</function></link>
 "rtcp_bye": we have received a RTCP bye packet. Arguments of the callback
             functions are a const char * containing the leaving reason and
             the user_data.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> 	a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>signal</parameter>&nbsp;:</term>
<listitem><simpara>		the name of a signal
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>cb</parameter>&nbsp;:</term>
<listitem><simpara>			a <link linkend="RtpCallback"><type>RtpCallback</type></link>
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>Param4</parameter>&nbsp;:</term>
<listitem><simpara>
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> 0 on success, -EOPNOTSUPP if the signal does not exists, -1 if no more callbacks
can be assigned to the signal type.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-signal-disconnect-by-callback" role="function">rtp_session_signal_disconnect_by_callback ()</title>
<indexterm><primary>rtp_session_signal_disconnect_by_callback</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_signal_disconnect_by_callback
                                            (<link linkend="RtpSession">RtpSession</link> *session,
                                             const <link linkend="char">char</link> *signal,
                                             <link linkend="RtpCallback">RtpCallback</link> cb);</programlisting>
<para>
Removes callback function <parameter>cb</parameter> to the list of callbacks for signal <parameter>signal</parameter>.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>signal</parameter>&nbsp;:</term>
<listitem><simpara>	a signal name
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>cb</parameter>&nbsp;:</term>
<listitem><simpara>		a callback function.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> 0 on success, -ENOENT if the callbacks was not found.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-send-with-ts" role="function">rtp_session_send_with_ts ()</title>
<indexterm><primary>rtp_session_send_with_ts</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_send_with_ts        (<link linkend="RtpSession">RtpSession</link> *session,
                                             const <link linkend="char">char</link> *buffer,
                                             <link linkend="int">int</link> len,
                                             <link linkend="uint32-t">uint32_t</link> userts);</programlisting>
<para>
Send a rtp datagram to the destination set by <link linkend="rtp-session-set-remote-addr"><function>rtp_session_set_remote_addr()</function></link> containing
the data from <parameter>buffer</parameter> with timestamp <parameter>userts</parameter>. This is a high level function that uses
<link linkend="rtp-session-create-packet"><function>rtp_session_create_packet()</function></link> and <link linkend="rtp-session-sendm-with-ts"><function>rtp_session_sendm_with_ts()</function></link> to send the data.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>buffer</parameter>&nbsp;:</term>
<listitem><simpara>	a buffer containing the data to be sent in a rtp packet.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>len</parameter>&nbsp;:</term>
<listitem><simpara>		the length of the data buffer, in bytes.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>userts</parameter>&nbsp;:</term>
<listitem><simpara>	the timestamp of the data to be sent. Refer to the rfc to know what it is.
</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-recv-with-ts" role="function">rtp_session_recv_with_ts ()</title>
<indexterm><primary>rtp_session_recv_with_ts</primary></indexterm><programlisting><link linkend="int">int</link>         rtp_session_recv_with_ts        (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="char">char</link> *buffer,
                                             <link linkend="int">int</link> len,
                                             <link linkend="uint32-t">uint32_t</link> time,
                                             <link linkend="int">int</link> *have_more);</programlisting>
<para>
Tries to read the bytes of the incoming rtp stream related to timestamp <parameter>time</parameter>. In case 
where the user supplied buffer <parameter>buffer</parameter> is not large enough to get all the data 
related to timestamp <parameter>time</parameter>, then *( <parameter>have_more</parameter>) is set to 1 to indicate that the application
should recall the function with the same timestamp to get more data.
</para>
<para>
 When the rtp session is scheduled (see <link linkend="rtp-session-set-scheduling-mode"><function>rtp_session_set_scheduling_mode()</function></link> ), and the 
blocking mode is on (see <link linkend="rtp-session-set-blocking-mode"><function>rtp_session_set_blocking_mode()</function></link> ), then the calling thread
is suspended until the timestamp given as argument expires, whatever a received packet 
fits the query or not.
</para>
<para>
Important note: it is clear that the application cannot know the timestamp of the first
packet of the incoming stream, because it can be random. The <parameter>time</parameter> timestamp given to the
function is used relatively to first timestamp of the stream. In simple words, 0 is a good
value to start calling this function.
</para>
<para>
This function internally calls <link linkend="rtp-session-recvm-with-ts"><function>rtp_session_recvm_with_ts()</function></link> to get a rtp packet. The content
of this packet is then copied into the user supplied buffer in an intelligent manner:
the function takes care of the size of the supplied buffer and the timestamp given in  
argument. Using this function it is possible to read continous audio data (e.g. pcma,pcmu...)
with for example a standart buffer of size of 160 with timestamp incrementing by 160 while the incoming
stream has a different packet size.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>
<listitem><simpara> a rtp session.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>buffer</parameter>&nbsp;:</term>
<listitem><simpara>	a user supplied buffer to write the data.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>len</parameter>&nbsp;:</term>
<listitem><simpara>		the length in bytes of the user supplied buffer.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>time</parameter>&nbsp;:</term>
<listitem><simpara>	the timestamp wanted.
</simpara></listitem></varlistentry>
<varlistentry><term><parameter>have_more</parameter>&nbsp;:</term>
<listitem><simpara> the address of an integer to indicate if more data is availlable for the given timestamp.
</simpara></listitem></varlistentry>
<varlistentry><term><emphasis>Returns</emphasis>&nbsp;:</term><listitem><simpara> if a packet was availlable with the corresponding timestamp supplied in argument 
then the number of bytes written in the user supplied buffer is returned. If no packets
are availlable, either because the sender has not started to send the stream, or either
because silence packet are not transmitted, or either because the packet was lost during
network transport, then the function returns zero.
</simpara></listitem></varlistentry>
</variablelist></refsect2>
<refsect2>
<title><anchor id="rtp-session-recvm-with-ts" role="function">rtp_session_recvm_with_ts ()</title>
<indexterm><primary>rtp_session_recvm_with_ts</primary></indexterm><programlisting><link linkend="mblk-t">mblk_t</link>*     rtp_session_recvm_with_ts       (<link linkend="RtpSession">RtpSession</link> *session,
                                             <link linkend="uint32-t">uint32_t</link> user_ts);</programlisting>
<para>
Try to get a rtp packet presented as a mblk_t structure from the rtp session.
The <parameter>user_ts</parameter> parameter is relative to the first timestamp of the incoming stream. In other
words, the application does not have to know the first timestamp of the stream, it can
simply call for the first time this function with <parameter>user_ts</parameter>=0, and then incrementing it
as it want. The RtpSession takes care of synchronisation between the stream timestamp
and the user timestamp given here.</para>
<para>

</para><variablelist role="params">
<varlistentry><term><parameter>session</parameter>&nbsp;:</term>