From 63170c4333b792805df5170e43d516b479a75421 Mon Sep 17 00:00:00 2001
From: markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>
Date: Sat, 28 Jun 2003 16:40:02 +0000
Subject: Add SIP/RTP video support, video enable app_echo, start on RTCP

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1128 f38db490-d61c-443f-a65b-d21fe96a405b
---
 channel.c | 20 +++++++++++++++++++-
 1 file changed, 19 insertions(+), 1 deletion(-)

(limited to 'channel.c')

diff --git a/channel.c b/channel.c
index 3071113c3..2127a46bd 100755
--- a/channel.c
+++ b/channel.c
@@ -1,4 +1,4 @@
-/*
+ /*
  * Asterisk -- A telephony toolkit for Linux.
  *
  * Channel Management
@@ -1269,6 +1269,17 @@ int ast_prod(struct ast_channel *chan)
 	return 0;
 }
 
+int ast_write_video(struct ast_channel *chan, struct ast_frame *fr)
+{
+	int res;
+	if (!chan->pvt->write_video)
+		return 0;
+	res = ast_write(chan, fr);
+	if (!res)
+		res = 1;
+	return res;
+}
+
 int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 {
 	int res = -1;
@@ -1306,6 +1317,13 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 		if (chan->pvt->send_text)
 			res = chan->pvt->send_text(chan, (char *) fr->data);
 		break;
+	case AST_FRAME_VIDEO:
+		/* XXX Handle translation of video codecs one day XXX */
+		if (chan->pvt->write_video)
+			res = chan->pvt->write_video(chan, fr);
+		else
+			res = 0;
+		break;
 	default:
 		if (chan->pvt->write) {
 			if (chan->pvt->writetrans) {
-- 
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