Merged revisions 224855 via svnmerge from

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

........
  r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines
  
  Pay attention to the return value of the manipulate function.
  While this looks like an optimization, it prevents a crash from occurring
  when used with certain audiohook callbacks (diagnosed with SVN trunk,
  backported to 1.4 to keep the source consistent across versions).
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2 files changed, 24 insertions, 12 deletions

diff --git a/funcs/func_speex.c b/funcs/func_speex.c
index d5246b0bd..699d4fd92 100644
--- a/funcs/func_speex.c
+++ b/funcs/func_speex.c
@@ -142,25 +142,24 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
 	struct ast_datastore *datastore = NULL;
 	struct speex_direction_info *sdi = NULL;
 	struct speex_info *si = NULL;
+	char source[80];
 
 	/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
 	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
-		return 0;
+		return -1;
 	}
-	
-	ast_channel_lock(chan);
+
+	/* We are called with chan already locked */
 	if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
-		ast_channel_unlock(chan);
-		return 0;
+		return -1;
 	}
-	ast_channel_unlock(chan);
 
 	si = datastore->data;
 
 	sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
 
 	if (!sdi) {
-		return 0;
+		return -1;
 	}
 
 	if (sdi->samples != frame->samples) {
@@ -171,7 +170,7 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
 		if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
 			return -1;
 		}
-		
+
 		speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
 
 		if (sdi->agc) {
@@ -182,6 +181,12 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
 	}
 
 	speex_preprocess(sdi->state, frame->data.ptr, NULL);
+	snprintf(source, sizeof(source), "%s/speex", frame->src);
+	if (frame->mallocd & AST_MALLOCD_SRC) {
+		ast_free((char *) frame->src);
+	}
+	frame->src = ast_strdup(source);
+	frame->mallocd |= AST_MALLOCD_SRC;
 
 	return 0;
 }
diff --git a/main/audiohook.c b/main/audiohook.c
index 1527a2d82..950b0eb2d 100644
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -574,7 +574,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 	struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
 	struct ast_audiohook *audiohook = NULL;
 	int samples = frame->samples;
-	
+
 	/* If the frame coming in is not signed linear we have to send it through the in_translate path */
 	if (frame->subclass != AST_FORMAT_SLINEAR) {
 		if (in_translate->format != frame->subclass) {
@@ -645,11 +645,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 				continue;
 			}
 			/* Feed in frame to manipulation */
-			audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
+			if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
+				ast_frfree(middle_frame);
+				middle_frame = NULL;
+			}
 			ast_audiohook_unlock(audiohook);
 		}
 		AST_LIST_TRAVERSE_SAFE_END;
-		end_frame = middle_frame;
+		if (middle_frame) {
+			end_frame = middle_frame;
+		}
 	}
 
 	/* Now we figure out what to do with our end frame (whether to transcode or not) */
@@ -677,7 +682,9 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 		}
 	} else {
 		/* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
-		ast_frfree(middle_frame);
+		if (middle_frame) {
+			ast_frfree(middle_frame);
+		}
 	}
 
 	return end_frame;