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Wrap rtp_engine.h header comments to 80 characters.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Sean Bright
2009-07-03 02:02:50 +00:00
parent a894c33cb3
commit c381cf82e7
1 changed files with 28 additions and 22 deletions
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include/asterisk/rtp_engine.h
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@@ -26,34 +26,40 @@
/*! /*!
* \page AstRTPEngine Asterisk RTP Engine API * \page AstRTPEngine Asterisk RTP Engine API
* *
* The purpose of this API is to provide a way for multiple RTP stacks to be used inside * The purpose of this API is to provide a way for multiple RTP stacks to be
* of Asterisk without any module that uses RTP knowing any different. To the module each RTP * used inside of Asterisk without any module that uses RTP knowing any
* stack behaves the same. * different. To the module each RTP stack behaves the same.
* *
* An RTP session is called an instance and is made up of a combination of codec information, * An RTP session is called an instance and is made up of a combination of codec
* RTP engine, RTP properties, and address information. An engine name may be passed in to explicitly * information, RTP engine, RTP properties, and address information. An engine
* choose an RTP stack to be used but a default one will be used if none is provided. An address to use * name may be passed in to explicitly choose an RTP stack to be used but a
* for RTP may also be provided but the underlying RTP engine may choose a different address depending on * default one will be used if none is provided. An address to use for RTP may
* it's configuration. * also be provided but the underlying RTP engine may choose a different address
* depending on it's configuration.
* *
* An RTP engine is the layer between the RTP engine core and the RTP stack itself. The RTP engine core provides * An RTP engine is the layer between the RTP engine core and the RTP stack
* a set of callbacks to do various things (such as write audio out) that the RTP engine has to have implemented. * itself. The RTP engine core provides a set of callbacks to do various things
* (such as write audio out) that the RTP engine has to have implemented.
* *
* Glue is what binds an RTP instance to a channel. It is used to retrieve RTP instance information when * Glue is what binds an RTP instance to a channel. It is used to retrieve RTP
* performing remote or local bridging and is used to have the channel driver tell the remote side to change * instance information when performing remote or local bridging and is used to
* destination of the RTP stream. * have the channel driver tell the remote side to change destination of the RTP
* stream.
* *
* Statistics from an RTP instance can be retrieved using the ast_rtp_instance_get_stats API call. This essentially * Statistics from an RTP instance can be retrieved using the
* asks the RTP engine in use to fill in a structure with the requested values. It is not required for an RTP engine * ast_rtp_instance_get_stats API call. This essentially asks the RTP engine in
* to support all statistic values. * use to fill in a structure with the requested values. It is not required for
* an RTP engine to support all statistic values.
* *
* Properties allow behavior of the RTP engine and RTP engine core to be changed. For example, there is a property named * Properties allow behavior of the RTP engine and RTP engine core to be
* AST_RTP_PROPERTY_NAT which is used to tell the RTP engine to enable symmetric RTP if it supports it. It is not required * changed. For example, there is a property named AST_RTP_PROPERTY_NAT which is
* for an RTP engine to support all properties. * used to tell the RTP engine to enable symmetric RTP if it supports it. It is
* not required for an RTP engine to support all properties.
* *
* Codec information is stored using a separate data structure which has it's own set of API calls to add/remove/retrieve * Codec information is stored using a separate data structure which has it's
* information. They are used by the module after an RTP instance is created so that payload information is available for * own set of API calls to add/remove/retrieve information. They are used by the
* the RTP engine. * module after an RTP instance is created so that payload information is
* available for the RTP engine.
*/ */
#ifndef _ASTERISK_RTP_ENGINE_H #ifndef _ASTERISK_RTP_ENGINE_H