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Add SIP/RTP video support, video enable app_echo, start on RTCP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -39,7 +39,7 @@ extern "C" {
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#define MAX_LANGUAGE 20
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#define AST_MAX_FDS 4
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#define AST_MAX_FDS 8
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struct ast_generator {
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void *(*alloc)(struct ast_channel *chan, void *params);
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@@ -72,7 +72,13 @@ struct ast_channel {
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int writeinterrupt;
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/*! Who are we bridged to, if we're bridged */
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struct ast_channel *bridge;
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struct ast_channel *bridge;
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/*! Who did we call? */
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struct ast_channel *dialed;
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/*! Who called us? */
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struct ast_channel *dialing;
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/*! Reverse the dialed link (0 false, 1 true) */
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int reversedialed;
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/*! Channel that will masquerade as us */
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struct ast_channel *masq;
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/*! Who we are masquerading as */
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@@ -122,7 +128,7 @@ struct ast_channel {
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/*! Number of rings so far */
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int rings;
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/*! Current level of application */
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int stack;
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int stack;
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/*! Kinds of data this channel can natively handle */
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@@ -199,7 +205,7 @@ struct ast_channel {
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unsigned int pickupgroup;
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/*! For easy linking */
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struct ast_channel *next;
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struct ast_channel *next;
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};
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@@ -452,7 +458,7 @@ int ast_waitfor_n_fd(int *fds, int n, int *ms, int *exception);
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//! Reads a frame
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/*!
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/*!
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* \param chan channel to read a frame from
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* Read a frame. Returns a frame, or NULL on error. If it returns NULL, you
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best just stop reading frames and assume the channel has been
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@@ -468,6 +474,15 @@ struct ast_frame *ast_read(struct ast_channel *chan);
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*/
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int ast_write(struct ast_channel *chan, struct ast_frame *frame);
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//! Write video frame to a channel
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/*!
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* \param chan destination channel of the frame
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* \param frame frame that will be written
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* This function writes the given frame to the indicated channel.
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* It returns 1 on success, 0 if not implemented, and -1 on failure.
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*/
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int ast_write_video(struct ast_channel *chan, struct ast_frame *frame);
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/* Send empty audio to prime a channel driver */
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int ast_prod(struct ast_channel *chan);
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