Update .version and ChangeLog

git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.6.0.27@260657 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.version

@@ -1 +1 @@
-1.6.0.27-rc2
+1.6.0.27

ChangeLog

@@ -1,3 +1,27 @@
+2010-05-03  Leif Madsen <lmadsen@digium.com>
+
+	* Asterisk 1.6.0.27 Released
+
+2010-04-29  Leif Madsen <lmadsen@digium.com>
+
+	* Astersik 1.6.0.27-rc3 Released
+
+2010-04-29 10:31 +0000 [r260053]  David Vossel <dvossel@digium.com>
+
+	* include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
+	  audiohook_write_list.  (closes issue 0017052) Reported by: dvossel
+	  Tested by: dvossel. (closes issue 0016196) Reported by: atis.
+	  Review: https://reviewboard.asterisk.org/r/623/
+
+2010-04-28 10:31 +0000 [r259936]  David Vossel <dvossel@digium.com>
+
+	* channels/chan_local.c, main/channel.c: Resolves deadlocks in
+	  chan_local.  (closes issue 0017185) Reported by: schmoozecom
+	  Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
+	  issue_17185_v2.diff uploaded by dvossel (license 671) Tested
+	  by: schmoozecom, GameGamer43
+	  Review: https://reviewboard.asterisk.org/r/631/
+
 2010-04-13  Leif Madsen <lmadsen@digium.com>
 
 	* Asterisk 1.6.0.27-rc2 Released

channels/chan_local.c

@@ -540,12 +540,12 @@ static int local_hangup(struct ast_channel *ast)
 			/* Deadlock avoidance */
 			while (p->owner && ast_channel_trylock(p->owner)) {
 				ast_mutex_unlock(&p->lock);
-				if (ast) {
-					ast_channel_unlock(ast);
+				if (p->chan) {
+					ast_channel_unlock(p->chan);
 				}
 				usleep(1);
-				if (ast) {
-					ast_channel_lock(ast);
+				if (p->chan) {
+					ast_channel_lock(p->chan);
 				}
 				ast_mutex_lock(&p->lock);
 			}
@@ -560,8 +560,17 @@ static int local_hangup(struct ast_channel *ast)
 	} else {
 		ast_module_user_remove(p->u_owner);
 		while (p->chan && ast_channel_trylock(p->chan)) {
-			DEADLOCK_AVOIDANCE(&p->lock);
+				ast_mutex_unlock(&p->lock);
+				if (p->owner) {
+					ast_channel_unlock(p->owner);
+				}
+				usleep(1);
+				if (p->owner) {
+					ast_channel_lock(p->owner);
+				}
+				ast_mutex_lock(&p->lock);
 		}
+
 		p->owner = NULL;
 		if (p->chan) {
 			ast_queue_hangup(p->chan);

include/asterisk/audiohook.h

@@ -73,9 +73,16 @@ struct ast_audiohook;
  * \param chan Channel
  * \param frame Frame of audio to manipulate
  * \param direction Direction frame came from
- * \return Returns 0 on success, -1 on failure
- * \note An audiohook does not have any reference to a private data structure for manipulate types. It is up to the manipulate callback to store this data
- *       via it's own method. An example would be datastores.
+ * \return Returns 0 on success, -1 on failure.
+ * \note An audiohook does not have any reference to a private data structure for manipulate
+ *       types. It is up to the manipulate callback to store this data via it's own method.
+ *       An example would be datastores.
+ * \note The input frame should never be freed or corrupted during a manipulate callback.
+ *       If the callback has the potential to corrupt the frame's data during manipulation,
+ *       local data should be used for the manipulation and only copied to the frame on
+ *       success.
+ * \note A failure return value indicates that the frame was not manipulated and that
+ *       is being returned in its original state.
  */
 typedef int (*ast_audiohook_manipulate_callback)(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction);
 

main/audiohook.c

@@ -572,7 +572,29 @@ static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, str
 	return frame;
 }
 
-/*! \brief Pass an AUDIO frame off to be handled by the audiohook core
+/*!
+ * \brief Pass an AUDIO frame off to be handled by the audiohook core
+ *
+ * \details
+ * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
+ * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
+ * input frame.
+ *
+ * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
+ *         format.  The result of this part is middle_frame is guaranteed to be in
+ *         SLINEAR format for Part_2.
+ * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
+ *         either a new frame as result of the translation, or points directly to the start_frame
+ *         because no translation to SLINEAR audio was required.  The result of this part
+ *         is end_frame will be updated to point to middle_frame if any audiohook manipulation
+ *         took place.
+ * Part_3: Translate end_frame's audio back into the format of start frame if necessary.
+ *         At this point if middle_frame != end_frame, we are guaranteed that no manipulation
+ *         took place and middle_frame can be freed as it was translated... If middle_frame was
+ *         not translated and still pointed to start_frame, it would be equal to end_frame as well
+ *         regardless if manipulation took place which would not result in this free.  The result
+ *         of this part is end_frame is guaranteed to be the format of start_frame for the return.
+ *         
  * \param chan Channel that the list is coming off of
  * \param audiohook_list List of audiohooks
  * \param direction Direction frame is coming in from
@@ -587,6 +609,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 	struct ast_audiohook *audiohook = NULL;
 	int samples = frame->samples;
 
+	/* ---Part_1. translate start_frame to SLINEAR if necessary. */
 	/* 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) {
@@ -601,6 +624,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 		samples = middle_frame->samples;
 	}
 
+	/* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
 	/* Queue up signed linear frame to each spy */
 	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
 		ast_audiohook_lock(audiohook);
@@ -654,20 +678,21 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 				audiohook->manipulate_callback(audiohook, chan, NULL, direction);
 				continue;
 			}
-			/* Feed in frame to manipulation */
+			/* Feed in frame to manipulation. */
 			if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
-				ast_frfree(middle_frame);
-				middle_frame = NULL;
+				/* XXX IGNORE FAILURE */
+
+				/* If the manipulation fails then the frame will be returned in its original state.
+				 * Since there are potentially more manipulator callbacks in the list, no action should
+				 * be taken here to exit early. */
 			}
 			ast_audiohook_unlock(audiohook);
 		}
 		AST_LIST_TRAVERSE_SAFE_END
-		if (middle_frame) {
-			end_frame = middle_frame;
-		}
+		end_frame = middle_frame;
 	}
 
-	/* Now we figure out what to do with our end frame (whether to transcode or not) */
+	/* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
 	if (middle_frame == end_frame) {
 		/* Middle frame was modified and became the end frame... let's see if we need to transcode */
 		if (end_frame->subclass != start_frame->subclass) {
@@ -692,9 +717,7 @@ 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 */
-		if (middle_frame) {
-			ast_frfree(middle_frame);
-		}
+		ast_frfree(middle_frame);
 	}
 
 	return end_frame;

main/channel.c

@@ -1986,25 +1986,22 @@ int ast_activate_generator(struct ast_channel *chan, struct ast_generator *gen,
 	int res = 0;
 
 	ast_channel_lock(chan);
-
 	if (chan->generatordata) {
 		if (chan->generator && chan->generator->release)
 			chan->generator->release(chan, chan->generatordata);
 		chan->generatordata = NULL;
 	}
-
-	ast_prod(chan);
 	if (gen->alloc && !(chan->generatordata = gen->alloc(chan, params))) {
 		res = -1;
 	}
-	
 	if (!res) {
 		ast_settimeout(chan, 160, generator_force, chan);
 		chan->generator = gen;
 	}
-
 	ast_channel_unlock(chan);
 
+	ast_prod(chan);
+
 	return res;
 }