Merged revisions 180372 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
  
  Fix problems when RTP packet frame size is changed
  
  During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
  
  This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
  
  Review: http://reviewboard.digium.com/r/184/
........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180373 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 41 insertions, 7 deletions

diff --git a/main/rtp.c b/main/rtp.c
index 54f3a85ff..3e24e25bb 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -3659,14 +3659,48 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
 
 void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
 {
-	int x;
-	for (x = 0; x < 32; x++) {  /* Ugly way */
-		rtp->pref.order[x] = prefs->order[x];
-		rtp->pref.framing[x] = prefs->framing[x];
+	struct ast_format_list current_format_old, current_format_new;
+
+	/* if no packets have been sent through this session yet, then
+	 *  changing preferences does not require any extra work
+	 */
+	if (rtp->lasttxformat == 0) {
+		rtp->pref = *prefs;
+		return;
 	}
-	if (rtp->smoother)
-		ast_smoother_free(rtp->smoother);
-	rtp->smoother = NULL;
+
+	current_format_old = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
+
+	rtp->pref = *prefs;
+
+	current_format_new = ast_codec_pref_getsize(&rtp->pref, rtp->lasttxformat);
+
+	/* if the framing desired for the current format has changed, we may have to create
+	 * or adjust the smoother for this session
+	 */
+	if ((current_format_new.inc_ms != 0) &&
+	    (current_format_new.cur_ms != current_format_old.cur_ms)) {
+		int new_size = (current_format_new.cur_ms * current_format_new.fr_len) / current_format_new.inc_ms;
+
+		if (rtp->smoother) {
+			ast_smoother_reconfigure(rtp->smoother, new_size);
+			if (option_debug) {
+				ast_log(LOG_DEBUG, "Adjusted smoother to %d ms and %d bytes\n", current_format_new.cur_ms, new_size);
+			}
+		} else {
+			if (!(rtp->smoother = ast_smoother_new(new_size))) {
+				ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
+				return;
+			}
+			if (current_format_new.flags) {
+				ast_smoother_set_flags(rtp->smoother, current_format_new.flags);
+			}
+			if (option_debug) {
+				ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", rtp->lasttxformat, current_format_new.cur_ms, new_size);
+			}
+		}
+	}
+
 }
 
 struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp)