From d564404d73d8def08d1962fbf0df12d4df90f617 Mon Sep 17 00:00:00 2001
From: Russell Bryant <russell@russellbryant.com>
Date: Wed, 5 Mar 2008 01:52:18 +0000
Subject: [PATCH] Fix a bug that I just noticed in the RTP code.  The
 calculation for setting the len field in an ast_frame of audio was wrong when
 G.722 is in use.  The len field represents the number of ms of audio that the
 frame contains.  It would have set the value to be twice what it should be.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 include/asterisk/frame.h | 11 +++++++++++
 main/rtp.c               |  2 +-
 main/translate.c         | 14 +++-----------
 3 files changed, 15 insertions(+), 12 deletions(-)

diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index fc3481136b..3f15430280 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -569,6 +569,17 @@ int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
  */
 int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2);
 
+/*!
+ * \brief Get the sample rate for a given format.
+ */
+static force_inline int ast_format_rate(int format)
+{
+	if (format == AST_FORMAT_G722)
+		return 16000;
+
+	return 8000;
+}
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif
diff --git a/main/rtp.c b/main/rtp.c
index cec29dc4af..6578c400e4 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1313,7 +1313,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
 		/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
 		ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
 		rtp->f.ts = timestamp / 8;
-		rtp->f.len = rtp->f.samples / 8;
+		rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
 	} else {
 		/* Video -- samples is # of samples vs. 90000 */
 		if (!rtp->lastividtimestamp)
diff --git a/main/translate.c b/main/translate.c
index 0aa21df122..3f5466d1fa 100644
--- a/main/translate.c
+++ b/main/translate.c
@@ -306,14 +306,6 @@ struct ast_trans_pvt *ast_translator_build_path(int dest, int source)
 	return head;
 }
 
-static inline int format_rate(int format)
-{
-	if (format == AST_FORMAT_G722)
-		return 16000;
-
-	return 8000;
-}
-
 /*! \brief do the actual translation */
 struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f, int consume)
 {
@@ -350,7 +342,7 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
 			path->nextout = f->delivery;
 		}
 		/* Predict next incoming sample */
-		path->nextin = ast_tvadd(path->nextin, ast_samp2tv(f->samples, format_rate(f->subclass)));
+		path->nextin = ast_tvadd(path->nextin, ast_samp2tv(f->samples, ast_format_rate(f->subclass)));
 	}
 	delivery = f->delivery;
 	for ( ; out && p ; p = p->next) {
@@ -374,7 +366,7 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
 		
 		/* Predict next outgoing timestamp from samples in this
 		   frame. */
-		path->nextout = ast_tvadd(path->nextout, ast_samp2tv(out->samples, format_rate(out->subclass)));
+		path->nextout = ast_tvadd(path->nextout, ast_samp2tv(out->samples, ast_format_rate(out->subclass)));
 	} else {
 		out->delivery = ast_tv(0, 0);
 		ast_set2_flag(out, has_timing_info, AST_FRFLAG_HAS_TIMING_INFO);
@@ -397,7 +389,7 @@ static void calc_cost(struct ast_translator *t, int seconds)
 	struct ast_trans_pvt *pvt;
 	struct timeval start;
 	int cost;
-	int out_rate = format_rate(t->dstfmt);
+	int out_rate = ast_format_rate(t->dstfmt);
 
 	if (!seconds)
 		seconds = 1;