From aada5c97a114d2e377532b205fb24f7e1052236c Mon Sep 17 00:00:00 2001
From: kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>
Date: Thu, 5 Mar 2009 18:36:31 +0000
Subject: Merged revisions 180373 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/trunk

................
  r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines

  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/branches/1.6.0@180377 f38db490-d61c-443f-a65b-d21fe96a405b
---
 main/rtp.c | 48 +++++++++++++++++++++++++++++++++++++++++-------
 1 file changed, 41 insertions(+), 7 deletions(-)

(limited to 'main/rtp.c')

diff --git a/main/rtp.c b/main/rtp.c
index 2c647e78b..a63dc282d 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -3124,14 +3124,48 @@ 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)
-- 
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