[asterisk-commits] dvossel: trunk r317197 - in /trunk: bridges/ funcs/

[SVN commits to the Asterisk project]

Author: dvossel
Date: Thu May  5 13:08:42 2011
New Revision: 317197

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=317197
Log:
Fixes reliability issues with func_jitterbuffer's usage in the new ConfBridge application.

Modified:
    trunk/bridges/bridge_softmix.c
    trunk/funcs/func_jitterbuffer.c

Modified: trunk/bridges/bridge_softmix.c
URL: http://svnview.digium.com/svn/asterisk/trunk/bridges/bridge_softmix.c?view=diff&rev=317197&r1=317196&r2=317197
==============================================================================
--- trunk/bridges/bridge_softmix.c (original)
+++ trunk/bridges/bridge_softmix.c Thu May  5 13:08:42 2011
@@ -429,15 +429,20 @@
 		bridge_channel->tech_args.silence_threshold :
 		DEFAULT_SOFTMIX_SILENCE_THRESHOLD;
 	char update_talking = -1;  /* if this is set to 0 or 1, tell the bridge that the channel has started or stopped talking. */
+	int res = AST_BRIDGE_WRITE_SUCCESS;
 
 	/* Only accept audio frames, all others are unsupported */
 	if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) {
 		softmix_pass_dtmf(bridge, bridge_channel, frame);
-		return AST_BRIDGE_WRITE_SUCCESS;
+		goto no_audio;
 	} else if (frame->frametype != AST_FRAME_VOICE) {
-		return AST_BRIDGE_WRITE_UNSUPPORTED;
-	}
-
+		res = AST_BRIDGE_WRITE_UNSUPPORTED;
+		goto no_audio;
+	} else if (frame->datalen == 0) {
+		goto no_audio;
+	}
+
+	/* If we made it here, we are going to write the frame into the conference */
 	ast_mutex_lock(&sc->lock);
 
 	ast_dsp_silence(sc->dsp, frame, &totalsilence);
@@ -480,7 +485,20 @@
 		ast_bridge_notify_talking(bridge, bridge_channel, update_talking);
 	}
 
-	return AST_BRIDGE_WRITE_SUCCESS;
+	return res;
+
+no_audio:
+	/* Even though the frame is not being written into the conference because it is not audio,
+	 * we should use this opportunity to check to see if a frame is ready to be written out from
+	 * the conference to the channel. */
+	ast_mutex_lock(&sc->lock);
+	if (sc->have_frame) {
+		ast_write(bridge_channel->chan, &sc->write_frame);
+		sc->have_frame = 0;
+	}
+	ast_mutex_unlock(&sc->lock);
+
+	return res;
 }
 
 /*! \brief Function called when the channel's thread is poked */

Modified: trunk/funcs/func_jitterbuffer.c
URL: http://svnview.digium.com/svn/asterisk/trunk/funcs/func_jitterbuffer.c?view=diff&rev=317197&r1=317196&r2=317197
==============================================================================
--- trunk/funcs/func_jitterbuffer.c (original)
+++ trunk/funcs/func_jitterbuffer.c Thu May  5 13:08:42 2011
@@ -203,6 +203,7 @@
 	struct jb_framedata *framedata = data;
 	struct timeval now_tv;
 	unsigned long now;
+	int putframe = 0; /* signifies if audio frame was placed into the buffer or not */
 
 	switch (event) {
 	case AST_FRAMEHOOK_EVENT_READ:
@@ -249,15 +250,31 @@
 		if (res == AST_JB_IMPL_OK) {
 			frame = &ast_null_frame;
 		}
+		putframe = 1;
 	}
 
 	if (frame->frametype == AST_FRAME_NULL) {
 		int res;
 		long next = framedata->jb_impl->next(framedata->jb_obj);
 
+		/* If now is earlier than the next expected output frame
+		 * from the jitterbuffer we may choose to pass on retrieving
+		 * a frame during this read iteration.  The only exception
+		 * to this rule is when an audio frame is placed into the buffer
+		 * and the time for the next frame to come out of the buffer is
+		 * at least within the timer_interval of the next output frame. By
+		 * doing this we are able to feed off the timing of the input frames
+		 * and only rely on our jitterbuffer timer when frames are dropped.
+		 * During testing, this hybrid form of timing gave more reliable results. */
 		if (now < next) {
-			return frame;
-		}
+			long int diff = next - now;
+			if (!putframe) {
+				return frame;
+			} else if (diff >= framedata->timer_interval) {
+				return frame;
+			}
+		}
+
 		res = framedata->jb_impl->get(framedata->jb_obj, &frame, now, framedata->timer_interval);
 		switch (res) {
 		case AST_JB_IMPL_OK: