diff --git a/src/add-ons/media/media-add-ons/mixer/AudioMixer.cpp b/src/add-ons/media/media-add-ons/mixer/AudioMixer.cpp
index 0be753f3fc..a44af22efd 100644
--- a/src/add-ons/media/media-add-ons/mixer/AudioMixer.cpp
+++ b/src/add-ons/media/media-add-ons/mixer/AudioMixer.cpp
@@ -21,6 +21,8 @@
 
 //
 
+#define SEND_NEW_BUFFER_EVENT 	(BTimedEventQueue::B_USER_EVENT + 1)
+
 mixer_input::mixer_input(media_input &input)
 {
 
@@ -225,7 +227,7 @@ AudioMixer::AudioMixer(BMediaAddOn *addOn)
 			fWeb(NULL),
 			fLatency(1),
 			fInternalLatency(1),
-			fStartTime(0), fNextEventTime(0),
+			fStartTime(0),
 			fFramesSent(0),
 			fOutputEnabled(true),
 			fBufferGroup(NULL),
@@ -236,27 +238,21 @@ AudioMixer::AudioMixer(BMediaAddOn *addOn)
 
 	BMediaNode::AddNodeKind(B_SYSTEM_MIXER);
 	
-	// set up the preferred output format
-	
+	// set up the preferred output format (although we will accept any format)
+	memset(&fPrefOutputFormat, 0, sizeof(fPrefOutputFormat)); // set everything to wildcard first
 	fPrefOutputFormat.type = B_MEDIA_RAW_AUDIO;
+	fPrefOutputFormat.u.raw_audio.byte_order = B_MEDIA_HOST_ENDIAN;
+	fPrefOutputFormat.u.raw_audio.channel_count = 2;
 	
-	fPrefOutputFormat.u.raw_audio.format = media_raw_audio_format::wildcard.format; // B_AUDIO_FLOAT
-	fPrefOutputFormat.u.raw_audio.frame_rate = media_raw_audio_format::wildcard.frame_rate; 
-	fPrefOutputFormat.u.raw_audio.byte_order = (B_HOST_IS_BENDIAN) ? B_MEDIA_BIG_ENDIAN : B_MEDIA_LITTLE_ENDIAN;
-
-	fPrefOutputFormat.u.raw_audio.buffer_size = media_raw_audio_format::wildcard.buffer_size; //2048
-	fPrefOutputFormat.u.raw_audio.channel_count = 2; //media_raw_audio_format::wildcard.channel_count;
-	
-	// set up the preferred input format
-	
+	// set up the preferred input format (although we will accept any format)
+	memset(&fPrefInputFormat, 0, sizeof(fPrefInputFormat)); // set everything to wildcard first
 	fPrefInputFormat.type = B_MEDIA_RAW_AUDIO;
-	
 	fPrefInputFormat.u.raw_audio.format = media_raw_audio_format::B_AUDIO_FLOAT;
 	fPrefInputFormat.u.raw_audio.frame_rate = 96000; //wildcard?
-	fPrefInputFormat.u.raw_audio.byte_order = (B_HOST_IS_BENDIAN) ? B_MEDIA_BIG_ENDIAN : B_MEDIA_LITTLE_ENDIAN;
+	fPrefInputFormat.u.raw_audio.byte_order = B_MEDIA_HOST_ENDIAN;
 
 	fPrefInputFormat.u.raw_audio.buffer_size = 1024;
-	fPrefInputFormat.u.raw_audio.channel_count = 2; //media_raw_audio_format::wildcard.channel_count;
+	fPrefInputFormat.u.raw_audio.channel_count = 2;
 
 	//fInput.source = media_source::null;
 
@@ -489,19 +485,16 @@ AudioMixer::HandleMessage( int32 message, const void *data, size_t size)
 status_t
 AudioMixer::AcceptFormat(const media_destination &dest, media_format *format)
 {
+	// check that the specified format is reasonable for the specified destination, and
+	// fill in any wildcard fields for which our BBufferConsumer has specific requirements. 
 
-	// we accept any raw audio 
-
-	if (IsValidDest(dest))
-	{	
-		if ((format->type != B_MEDIA_UNKNOWN_TYPE) && (format->type != B_MEDIA_RAW_AUDIO))
-			return B_MEDIA_BAD_FORMAT;
-		else
-			return B_OK;
-	}
-	else
+	if (!IsValidDest(dest))
 		return B_MEDIA_BAD_DESTINATION;	
-	
+
+	if ((format->type != B_MEDIA_UNKNOWN_TYPE) && (format->type != B_MEDIA_RAW_AUDIO))
+		return B_MEDIA_BAD_FORMAT;
+	else
+		return B_OK;
 }
 
 status_t
@@ -523,94 +516,116 @@ AudioMixer::GetNextInput(int32 *cookie, media_input *out_input)
 void
 AudioMixer::DisposeInputCookie(int32 cookie)
 {
-
 	// nothing yet
-
 }
 
 void
 AudioMixer::BufferReceived(BBuffer *buffer)
 {
-
-	if (buffer)
-	{
-		if (buffer->Header()->type == B_MEDIA_PARAMETERS) 
-		{
-			printf("Control Buffer Received\n");
-			ApplyParameterData(buffer->Data(), buffer->SizeUsed());
-		}
-		else 
-		{
-		
-			media_header *hdr = buffer->Header();
-					
-			// check input
-			
-			mixer_input *channel;
-			
-			int inputcount = fMixerInputs.CountItems();
-			
-			for (int i = 0; i < inputcount; i++)
-			{
-			
-				channel = (mixer_input *)fMixerInputs.ItemAt(i);
-				
-				if (channel->fInput.destination.id == hdr->destination)
-				{
-				
-					i = inputcount;
-							
-					bigtime_t now = TimeSource()->Now();
-					bigtime_t perf_time = hdr->start_time;				
-					bigtime_t how_early = perf_time - fLatency - now;
-		
-					if ((RunMode() != B_OFFLINE) && 
-					(RunMode() != B_RECORDING) &&
-					(how_early < 0))
-					{
-						printf("Received buffer %d usecs late from %s\n", -how_early, channel->fInput.name);
-						NotifyLateProducer(channel->fInput.source, -how_early, perf_time);
-					}
-					else
-					{
-					
-						size_t sample_size = channel->fInput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK;
-		
-						// calculate total byte offset for writing to the ringbuffer
-						// this takes account of the Event offset, as well as the buffer's start time			
-		
-						size_t total_offset = int(channel->fEventOffset + ((((perf_time - fNextEventTime) / 1000000) * 
-							channel->fInput.format.u.raw_audio.frame_rate) * 
-							sample_size * channel->fInput.format.u.raw_audio.channel_count)) % int(channel->fDataSize);
-		
-						char *indata = (char *)buffer->Data();
-														
-						if (buffer->SizeUsed() > (channel->fDataSize - total_offset))
-						{
-							memcpy(channel->fData + total_offset, indata, channel->fDataSize - total_offset);
-							memcpy(channel->fData, indata + (channel->fDataSize - total_offset), buffer->SizeUsed() - (channel->fDataSize - total_offset));	
-						} 
-						else
-							memcpy(channel->fData + total_offset, indata, buffer->SizeUsed());
-												
-					}
-				
-				}	
-			
-			if ((B_OFFLINE == RunMode()) && (B_DATA_AVAILABLE == channel->fProducerDataStatus))
-			{
-				RequestAdditionalBuffer(channel->fInput.source, buffer);
-			}
-			
-		}
-				
+	if (buffer->Header()->type == B_MEDIA_PARAMETERS) {
+		printf("Control Buffer Received\n");
+		ApplyParameterData(buffer->Data(), buffer->SizeUsed());
+		buffer->Recycle();
+		return;
 	}
 	
-	buffer->Recycle();
+	// to receive the buffer at the right time,
+	// push it through the event looper
+	media_timed_event event(buffer->Header()->start_time,
+							BTimedEventQueue::B_HANDLE_BUFFER,
+							buffer,
+							BTimedEventQueue::B_RECYCLE_BUFFER);
+	EventQueue()->AddEvent(event);
+}
+
+
+void
+AudioMixer::HandleInputBuffer(BBuffer *buffer)
+{
+	media_header *hdr = buffer->Header();
+
+	// check input
+	int inputcount = fMixerInputs.CountItems();
 	
+	for (int i = 0; i < inputcount; i++) {
+	
+		mixer_input *channel = (mixer_input *)fMixerInputs.ItemAt(i);
+				
+		if (channel->fInput.destination.id != hdr->destination)
+			continue;
+
+		bigtime_t now = TimeSource()->Now();
+		bigtime_t perf_time = hdr->start_time;				
+		bigtime_t how_late = now - perf_time;
+		bigtime_t event_latency = EventLatency();
+		
+		if (how_late > (event_latency + 2000)) {
+			how_late -= event_latency;
+			printf("Received buffer %Ld usecs late from %s\n", how_late, channel->fInput.name);
+			if (RunMode() != B_OFFLINE && RunMode() != B_RECORDING) {
+				printf("sending notify\n");
+				NotifyLateProducer(channel->fInput.source, max_c(500, how_late), perf_time);
+			} else if (RunMode() == B_DROP_DATA) {
+				printf("dropping buffer\n");
+				return;
+			}
+		}
+	
+		size_t sample_size = channel->fInput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK;
+	
+		// calculate total byte offset for writing to the ringbuffer
+		// this takes account of the Event offset, as well as the buffer's start time			
+/*		
+		size_t total_offset = int(channel->fEventOffset + ((((perf_time - fNextEventTime) / 1000000) * 
+				channel->fInput.format.u.raw_audio.frame_rate) * 
+				sample_size * channel->fInput.format.u.raw_audio.channel_count)) % int(channel->fDataSize);
+*/
+		size_t total_offset = int(channel->fEventOffset + channel->fInput.format.u.raw_audio.buffer_size)  % int(channel->fDataSize);
+		
+		char *indata = (char *)buffer->Data();
+														
+		if (buffer->SizeUsed() > (channel->fDataSize - total_offset))
+		{
+			memcpy(channel->fData + total_offset, indata, channel->fDataSize - total_offset);
+			memcpy(channel->fData, indata + (channel->fDataSize - total_offset), buffer->SizeUsed() - (channel->fDataSize - total_offset));	
+		} 
+		else
+			memcpy(channel->fData + total_offset, indata, buffer->SizeUsed());
+
+		if ((B_OFFLINE == RunMode()) && (B_DATA_AVAILABLE == channel->fProducerDataStatus))
+		{
+			RequestAdditionalBuffer(channel->fInput.source, buffer);
+		}
+		
+		break;
 	}
 }
 
+void
+AudioMixer::SendNewBuffer(bigtime_t event_time)
+{
+
+	BBuffer *outbuffer = fBufferGroup->RequestBuffer(fOutput.format.u.raw_audio.buffer_size, BufferDuration());
+	if (!outbuffer) {
+		printf("Could not allocate buffer\n");
+		return;
+	}
+				
+	FillMixBuffer(outbuffer->Data(), outbuffer->SizeAvailable());
+					
+	media_header *outheader = outbuffer->Header();
+	outheader->type = B_MEDIA_RAW_AUDIO;
+	outheader->size_used = fOutput.format.u.raw_audio.buffer_size;
+	outheader->time_source = TimeSource()->ID();
+	outheader->start_time = event_time;
+	
+	if (B_OK != SendBuffer(outbuffer, fOutput.destination)) {
+		printf("Could not send buffer to output : %s\n", fOutput.name);
+		outbuffer->Recycle();
+	}
+}	
+
+
 void
 AudioMixer::ProducerDataStatus( const media_destination &for_whom, 
 								int32 status, bigtime_t at_performance_time)
@@ -651,8 +666,14 @@ status_t
 AudioMixer::Connected( const media_source &producer, const media_destination &where,
 						const media_format &with_format, media_input *out_input)
 {
-	if (! IsValidDest(where))
+	if (!IsValidDest(where))
 		return B_MEDIA_BAD_DESTINATION;
+		
+	// We need to make sure that the outInput's name field contains a valid name,
+	// the name given the connection by the producer may still be an empty string.
+
+	// If we want the producer to use a specific BBufferGroup, we now need to call
+	// BMediaRoster::SetOutputBuffersFor() here to set the producer's buffer group
 	
 	char *name = out_input->name;
 	mixer_input *mixerInput;
@@ -997,7 +1018,7 @@ AudioMixer::PrepareToConnect(const media_source &what, const media_destination &
 		format->u.raw_audio.channel_count = 2;
 
 	if(format->u.raw_audio.byte_order == media_raw_audio_format::wildcard.byte_order)
-		format->u.raw_audio.byte_order = fPrefOutputFormat.u.raw_audio.byte_order;
+		format->u.raw_audio.byte_order = B_MEDIA_HOST_ENDIAN;
 
 	if (format->u.raw_audio.buffer_size == media_raw_audio_format::wildcard.buffer_size)
 		format->u.raw_audio.buffer_size = 1024;		// pick something comfortable to suggest
@@ -1038,6 +1059,7 @@ AudioMixer::Connect( status_t error, const media_source &source, const media_des
 	// Do so, then make sure we get our events early enough.
 	media_node_id id;
 	FindLatencyFor(fOutput.destination, &fLatency, &id);
+	printf("Downstream Latency is %Ld usecs\n", fLatency);
 
 	// we need at least the length of a full output buffer's latency (I think?)
 
@@ -1054,30 +1076,27 @@ AudioMixer::Connect( status_t error, const media_source &source, const media_des
 	bigtime_t latency_end = TimeSource()->RealTime();
 	
 	fInternalLatency = latency_end - latency_start;
-	printf("Latency set at %d usecs\n", fInternalLatency);
+	printf("Internal latency is %Ld usecs\n", fInternalLatency);
 	
 	delete mouse;
 	
-
 	// might need to tweak the latency
-
 	SetEventLatency(fLatency + fInternalLatency);
 
-	// reset our buffer duration, etc. to avoid later calculations
-	// crashes w/ divide-by-zero when connecting to a variety of nodes... 
-//	if (fOutput.format.u.raw_audio.frame_rate == media_raw_audio_format::wildcard.frame_rate)
-//	{
-//		fOutput.format.u.raw_audio.frame_rate = 44100;
-//	}
-	
-	bigtime_t duration = bigtime_t(1000000) * framesPerBuffer / bigtime_t(fOutput.format.u.raw_audio.frame_rate);
-	SetBufferDuration(duration);
+	// calculate buffer duration and set it	
+	if (fOutput.format.u.raw_audio.frame_rate == 0) {
+		// XXX must be adjusted later when the format is known
+		SetBufferDuration((framesPerBuffer * 1000000LL) / 44100);
+	} else {
+		SetBufferDuration((framesPerBuffer * 1000000LL) / fOutput.format.u.raw_audio.frame_rate);
+	}
 
 	// Set up the buffer group for our connection, as long as nobody handed us a
 	// buffer group (via SetBufferGroup()) prior to this.  That can happen, for example,
 	// if the consumer calls SetOutputBuffersFor() on us from within its Connected()
 	// method.
-	if (!fBufferGroup) AllocateBuffers();
+	if (!fBufferGroup)
+		AllocateBuffers();
 }
 
 
@@ -1101,19 +1120,19 @@ AudioMixer::Disconnect(const media_source &what, const media_destination &where)
 void 
 AudioMixer::LateNoticeReceived(const media_source &what, bigtime_t how_much, bigtime_t performance_time)
 {
-
 	// We've produced some late buffers... Increase Latency 
 	// is the only runmode in which we can do anything about this
+
+	printf("AudioMixer::LateNoticeReceived, %Ld too late at %Ld\n", how_much, performance_time);
 	
-	if (what == fOutput.source)
-		if (RunMode() == B_INCREASE_LATENCY)
-		{
+	if (what == fOutput.source) {
+		if (RunMode() == B_INCREASE_LATENCY) {
 			fInternalLatency += how_much;
+			
+			printf("AudioMixer: increasing internal latency to %Ld usec\n", fInternalLatency);
 			SetEventLatency(fLatency + fInternalLatency);
 		}
-		
-	printf("Late notice received. Buffer was %d usecs late\n", how_much);
-
+	}
 }
 
 
@@ -1172,91 +1191,42 @@ AudioMixer::NodeRegistered()
 
 
 void 
-AudioMixer::HandleEvent( const media_timed_event *event, bigtime_t lateness, 
-								bool realTimeEvent = false)
+AudioMixer::HandleEvent( const media_timed_event *event, bigtime_t lateness, bool realTimeEvent)
 {
-
 	switch (event->type)
 	{
 	
 		case BTimedEventQueue::B_HANDLE_BUFFER:
 		{
+			HandleInputBuffer((BBuffer *)event->pointer);
+			((BBuffer *)event->pointer)->Recycle();
+			break;
+		}
 
-			// check output
-			if (fOutput.destination != media_destination::null && fOutputEnabled) // this is in the wrong order too
-			{
-					
-				BBuffer *outbuffer = fBufferGroup->RequestBuffer(fOutput.format.u.raw_audio.buffer_size, BufferDuration());
-				
-				if (outbuffer)
-				{
+		case SEND_NEW_BUFFER_EVENT:
+		{
 
-					FillMixBuffer(outbuffer->Data(), outbuffer->SizeAvailable());
-					
-					media_header *outheader = outbuffer->Header();
-					outheader->type = B_MEDIA_RAW_AUDIO;
-					outheader->size_used = fOutput.format.u.raw_audio.buffer_size;
-					outheader->time_source = TimeSource()->ID();
-					
-					// if this is the first buffer, mark with the start time
-					// we need this to calculate the other buffer times
-					
-					if (fStartTime == 0) {
-						fStartTime = event->event_time;
-						fNextEventTime = fStartTime;
-					}
-
-					bigtime_t stamp;
-					if (RunMode() == B_RECORDING)
-						stamp = event->event_time; // this is actually the same as the other modes, since we're using 
-					else							// a timedevent queue and adding events at fNextEventTime
-					{
-						
-						// we're in a live performance mode
-						// use the start time we calculate at the end of the the mix loop
-						// this time is based on the offset of all media produced so far, 
-						// plus fStartTime, which is the recorded time of our first event
-
-						stamp = fNextEventTime;
-
-					}
-								
-					outheader->start_time = stamp;
-
-					status_t err = SendBuffer(outbuffer, fOutput.destination);
-												
-					if(err != B_OK)
-					{
-						outbuffer->Recycle();
-						printf("Could not send buffer to output : %s\n", fOutput.name);
-					}
-						
-					}
-				
-				else
-					printf("Failed to allocate a buffer\n");
-					
-				// even if we didn't get a buffer allocated, we still need to send the next event
-				
-				size_t sample_size = fOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK;
-	
-				int framesperbuffer = (fOutput.format.u.raw_audio.buffer_size / (sample_size * fOutput.format.u.raw_audio.channel_count));
-					
-				fFramesSent += (framesperbuffer);
-					
-				// calculate the start time for the next event
-					
-				fNextEventTime = bigtime_t(fStartTime + double(fFramesSent / fOutput.format.u.raw_audio.frame_rate) * 1000000.0);
-					
-				media_timed_event nextBufferEvent(fNextEventTime, BTimedEventQueue::B_HANDLE_BUFFER);
-				EventQueue()->AddEvent(nextBufferEvent);
-
-			}
-
-		
-		break;
+			if (fOutputEnabled && fOutput.destination != media_destination::null)
+				SendNewBuffer(event->event_time);
 			
-		}	
+			// if this is the first buffer, mark with the start time
+			// we need this to calculate the other buffer times
+			if (fStartTime == 0) {
+				fStartTime = event->event_time;
+			}
+			
+			// count frames that have been played
+			size_t sample_size = fOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK;
+			int framesperbuffer = (fOutput.format.u.raw_audio.buffer_size / (sample_size * fOutput.format.u.raw_audio.channel_count));
+
+			fFramesSent += framesperbuffer;
+			
+			// calculate the start time for the next event and add the event
+			bigtime_t nextevent = bigtime_t(fStartTime + double(fFramesSent / fOutput.format.u.raw_audio.frame_rate) * 1000000.0);
+			media_timed_event nextBufferEvent(nextevent, SEND_NEW_BUFFER_EVENT);
+			EventQueue()->AddEvent(nextBufferEvent);
+			break;
+		}
 		
 		case BTimedEventQueue::B_START:
 
@@ -1269,7 +1239,7 @@ AudioMixer::HandleEvent( const media_timed_event *event, bigtime_t lateness,
 				
 				//fThread = spawn_thread(_mix_thread_, "audio mixer thread", B_REAL_TIME_PRIORITY, this);
 				
-				media_timed_event firstBufferEvent(event->event_time, BTimedEventQueue::B_HANDLE_BUFFER);
+				media_timed_event firstBufferEvent(event->event_time, SEND_NEW_BUFFER_EVENT);
 
 				// Alternatively, we could call HandleEvent() directly with this event, to avoid a trip through
 				// the event queue, like this:
@@ -1289,6 +1259,7 @@ AudioMixer::HandleEvent( const media_timed_event *event, bigtime_t lateness,
 			// stopped - don't process any more buffers, flush all buffers from eventqueue
 
 			EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true, BTimedEventQueue::B_HANDLE_BUFFER);
+			EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true, SEND_NEW_BUFFER_EVENT);
 			break;
 		}
 
@@ -1344,9 +1315,14 @@ AudioMixer::AllocateBuffers()
 	// allocate enough buffers to span our downstream latency, plus one
 	size_t size = fOutput.format.u.raw_audio.buffer_size;
 	int32 count = int32((fLatency / (BufferDuration() + 1)) + 1);
-
-	fBufferGroup = new BBufferGroup(size, count);
 	
+	if (count < 2) {
+		printf("AudioMixer: calculated only %ld buffers, that's not enough\n", count);
+		count = 2;
+	}
+	
+	printf("AudioMixer: allocating %ld buffers\n", count);
+	fBufferGroup = new BBufferGroup(size, count);
 }
 
 // use this later for separate threads
diff --git a/src/add-ons/media/media-add-ons/mixer/AudioMixer.h b/src/add-ons/media/media-add-ons/mixer/AudioMixer.h
index 941fa4a5ea..55184bcbc2 100644
--- a/src/add-ons/media/media-add-ons/mixer/AudioMixer.h
+++ b/src/add-ons/media/media-add-ons/mixer/AudioMixer.h
@@ -47,6 +47,9 @@ class AudioMixer :
 		
 		void 				AllocateBuffers();
 		status_t			FillMixBuffer(void *outbuffer, size_t size);
+		
+		void				SendNewBuffer(bigtime_t event_time);
+		void				HandleInputBuffer(BBuffer *buffer);
 					
 	// BMediaNode methods
 
@@ -200,8 +203,8 @@ class AudioMixer :
 		BMediaAddOn		*	fAddOn;
 		BParameterWeb	*	fWeb; // local pointer to parameterweb
 
-		bigtime_t 			fLatency, fInternalLatency; // latency (total and internal)
-		bigtime_t			fStartTime, fNextEventTime; // time node started, time of next (output) event
+		bigtime_t 			fLatency, fInternalLatency; // latency (downstream and internal)
+		bigtime_t			fStartTime; // time node started
 		uint64 				fFramesSent; // audio frames sent
 		bool				fOutputEnabled;
 		
diff --git a/src/add-ons/media/media-add-ons/multi_audio/MultiAudioNode.cpp b/src/add-ons/media/media-add-ons/multi_audio/MultiAudioNode.cpp
index 008f1d1c1e..4b4345e595 100644
--- a/src/add-ons/media/media-add-ons/multi_audio/MultiAudioNode.cpp
+++ b/src/add-ons/media/media-add-ons/multi_audio/MultiAudioNode.cpp
@@ -130,6 +130,7 @@ MultiAudioNode::MultiAudioNode(BMediaAddOn *addon, char* name, MultiAudioDevice
 	AddNodeKind( B_PHYSICAL_INPUT );
 		
 	// initialize our preferred format object
+	memset(&fPreferredFormat, 0, sizeof(fPreferredFormat)); // set everything to wildcard first
 	fPreferredFormat.type = B_MEDIA_RAW_AUDIO;
 	fPreferredFormat.u.raw_audio.format = MultiAudioDevice::convert_multiaudio_format_to_media_format(fDevice->MFI.output.format);
 	fPreferredFormat.u.raw_audio.channel_count = 2;
diff --git a/src/kits/media/MediaDefs.cpp b/src/kits/media/MediaDefs.cpp
index ffd09728a7..ae1c361357 100644
--- a/src/kits/media/MediaDefs.cpp
+++ b/src/kits/media/MediaDefs.cpp
@@ -176,9 +176,9 @@ media_multistream_format media_multistream_format::wildcard = {0};
  *************************************************************/
 
 bool
-media_format::Matches(const media_format *otherFormat) const
+media_format::Matches(const media_format *other) const
 {
-	return format_is_compatible(*this, *otherFormat);
+	return type == other->type; // XXX fixthis
 }
 
 
@@ -412,10 +412,7 @@ bool operator==(const media_format & a, const media_format & b)
 /* return true if a and b are compatible (accounting for wildcards) */
 bool format_is_compatible(const media_format & a, const media_format & b)	/* a is the format you want to feed to something accepting b */
 {
-	UNIMPLEMENTED();
-	if (a.type == b.type)
-		return true;
-	return false;
+	return a.Matches(&b);
 }
 
 bool string_for_format(const media_format & f, char * buf, size_t size)
diff --git a/src/kits/media/MediaEventLooper.cpp b/src/kits/media/MediaEventLooper.cpp
index f8c8735c9f..12bc12eb3c 100644
--- a/src/kits/media/MediaEventLooper.cpp
+++ b/src/kits/media/MediaEventLooper.cpp
@@ -33,6 +33,7 @@ static char __copyright[] = "Copyright (c) 2002, 2003 Marcus Overhagen <Marcus@O
 #include <MediaEventLooper.h>
 #include <TimeSource.h>
 #include <scheduler.h>
+#include <Buffer.h>
 #include "debug.h"
 
 // XXX The bebook says that the latency is always calculated in realtime
@@ -177,6 +178,7 @@ BMediaEventLooper::SetRunMode(run_mode mode)
 		if(fControlThread > 0) {
 			set_thread_priority(fControlThread, fCurrentPriority);
 			fSchedulingLatency = estimate_max_scheduling_latency(fControlThread);
+			printf("BMediaEventLooper: SchedulingLatency is %Ld\n", fSchedulingLatency);
 		}
 	}
 
@@ -221,7 +223,6 @@ BMediaEventLooper::ControlLoop()
 			// BMediaEventLooper compensates your performance time by adding the event latency
 			// (see SetEventLatency()) and the scheduling latency (or, for real-time events, 
 			// only the scheduling latency). 
-			// latency = fOut.downstream_latency + fOut.processing_latency + fSchedulingLatency;
 			// XXX well, fix this later
 			latency = fEventLatency + fSchedulingLatency; 
 			
@@ -261,10 +262,10 @@ BMediaEventLooper::ControlLoop()
 		if (err == B_OK) {
 			bigtime_t lateness;
 			if (is_realtime)
-				lateness = TimeSource()->RealTime() - event.event_time;
+				lateness = TimeSource()->RealTime() - event.event_time - fEventLatency;
 			else
-				lateness = TimeSource()->Now() - event.event_time;
-			DispatchEvent(&event,lateness,is_realtime);
+				lateness = TimeSource()->Now() - event.event_time - fEventLatency;
+			DispatchEvent(&event, lateness, is_realtime);
 		}
 	}
 }
@@ -356,6 +357,7 @@ BMediaEventLooper::SetPriority(int32 priority)
 	if(fControlThread > 0) {
 		set_thread_priority(fControlThread, fCurrentPriority);
 		fSchedulingLatency = estimate_max_scheduling_latency(fControlThread);
+		printf("BMediaEventLooper: SchedulingLatency is %Ld\n", fSchedulingLatency);
 	}
 
 	return B_OK;
@@ -419,6 +421,7 @@ BMediaEventLooper::Run()
 
 	// get latency information
 	fSchedulingLatency = estimate_max_scheduling_latency(fControlThread);
+	printf("BMediaEventLooper: SchedulingLatency is %Ld\n", fSchedulingLatency);
 }
 
 
@@ -471,7 +474,8 @@ BMediaEventLooper::DispatchEvent(const media_timed_event *event,
 		default:
 			break;
 	}
-
+	
+	_DispatchCleanUp(event);
 }
 
 /*************************************************************
diff --git a/src/kits/media/MediaRoster.cpp b/src/kits/media/MediaRoster.cpp
index c72134f659..4ddadc4f1c 100644
--- a/src/kits/media/MediaRoster.cpp
+++ b/src/kits/media/MediaRoster.cpp
@@ -2109,6 +2109,8 @@ BMediaRosterEx::GetDormantFlavorInfo(media_addon_id addonid,
 									  dormant_flavor_info * out_flavor)
 {
 	CALLED();
+	if (out_flavor == NULL)
+		return B_BAD_VALUE;
 	
 	xfer_server_get_dormant_flavor_info msg;
 	xfer_server_get_dormant_flavor_info_reply *reply;
@@ -2300,8 +2302,27 @@ BMediaRoster::GetFormatFor(const media_output & output,
 						   media_format * io_format,
 						   uint32 flags)
 {
-	UNIMPLEMENTED();
-	return B_ERROR;
+	CALLED();
+	if (io_format == NULL)
+		return B_BAD_VALUE;
+	if ((output.node.kind & B_BUFFER_PRODUCER) == 0)
+		return B_MEDIA_BAD_NODE;
+	if (IS_INVALID_SOURCE(output.source))
+		return B_MEDIA_BAD_SOURCE;
+		
+	producer_format_suggestion_requested_request request;
+	producer_format_suggestion_requested_reply reply;
+	status_t rv;
+	
+	request.type = B_MEDIA_UNKNOWN_TYPE;
+	request.quality = 0; // XXX what should this be?
+		
+	rv = QueryPort(output.source.port, PRODUCER_FORMAT_SUGGESTION_REQUESTED, &request, sizeof(request), &reply, sizeof(reply));
+	if (rv != B_OK)
+		return rv;
+
+	*io_format = reply.format;
+	return B_OK;
 }
 
 						   
@@ -2310,8 +2331,27 @@ BMediaRoster::GetFormatFor(const media_input & input,
 						   media_format * io_format,
 						   uint32 flags)
 {
-	UNIMPLEMENTED();
-	return B_ERROR;
+	CALLED();
+	if (io_format == NULL)
+		return B_BAD_VALUE;
+	if ((input.node.kind & B_BUFFER_CONSUMER) == 0)
+		return B_MEDIA_BAD_NODE;
+	if (IS_INVALID_DESTINATION(input.destination))
+		return B_MEDIA_BAD_DESTINATION;
+		
+	consumer_accept_format_request request;
+	consumer_accept_format_reply reply;
+	status_t rv;
+	
+	request.dest = input.destination;
+	memset(&request.format, 0, sizeof(request.format)); // wildcard
+
+	rv = QueryPort(input.destination.port, CONSUMER_ACCEPT_FORMAT, &request, sizeof(request), &reply, sizeof(reply));
+	if (rv != B_OK)
+		return rv;
+
+	*io_format = reply.format;
+	return B_OK;
 }
 
 
@@ -2321,6 +2361,14 @@ BMediaRoster::GetFormatFor(const media_node & node,
 						   float quality)
 {
 	UNIMPLEMENTED();
+	if (io_format == NULL)
+		return B_BAD_VALUE;
+	if (IS_INVALID_NODE(node))
+		return B_MEDIA_BAD_NODE;
+	if ((node.kind & (B_BUFFER_CONSUMER | B_BUFFER_PRODUCER)) == 0)
+		return B_MEDIA_BAD_NODE;
+
+
 	return B_ERROR;
 }
 
diff --git a/src/kits/media/SharedBufferList.cpp b/src/kits/media/SharedBufferList.cpp
index 1caa75f71d..fa058f6f1b 100644
--- a/src/kits/media/SharedBufferList.cpp
+++ b/src/kits/media/SharedBufferList.cpp
@@ -269,6 +269,7 @@ _shared_buffer_list::RecycleBuffer(BBuffer *buffer)
 			reclaimed_count++;
 			if (info[i].reclaimed) {
 				FATAL("_shared_buffer_list: Error, BBuffer %p, id = %ld already reclaimed\n", buffer, id);
+				DEBUG_ONLY(debugger("buffer already reclaimed"));
 				continue;
 			}
 			info[i].reclaimed = true;
diff --git a/src/kits/media/SoundPlayNode.cpp b/src/kits/media/SoundPlayNode.cpp
index 5a1d5cdbe2..df43310bae 100644
--- a/src/kits/media/SoundPlayNode.cpp
+++ b/src/kits/media/SoundPlayNode.cpp
@@ -31,7 +31,8 @@ _SoundPlayNode::_SoundPlayNode(const char *name, const media_multi_audio_format
 	mInitCheckStatus(B_OK),
 	mOutputEnabled(true),
 	mBufferGroup(NULL),	
-	mFramesSent(0)
+	mFramesSent(0),
+	mTooEarlyCount(0)
 {
 	CALLED();
 	mPreferredFormat.type = B_MEDIA_RAW_AUDIO;
@@ -369,7 +370,7 @@ _SoundPlayNode::Connect(status_t error, const media_source& source, const media_
 	FindLatencyFor(mOutput.destination, &mLatency, &id);
 	fprintf(stderr, "\tdownstream latency = %Ld\n", mLatency);
 
-	mInternalLatency = 10LL;
+	mInternalLatency = 5000LL;
 	fprintf(stderr, "\tbuffer-filling took %Ld usec on this machine\n", mInternalLatency);
 	SetEventLatency(mLatency + mInternalLatency);
 
@@ -421,6 +422,8 @@ _SoundPlayNode::LateNoticeReceived(const media_source& what, bigtime_t how_much,
 {
 	CALLED();
 	
+	printf("_SoundPlayNode::LateNoticeReceived, %Ld too late at %Ld\n", how_much, performance_time);
+	
 	// is this our output?
 	if (what != mOutput.source)
 	{
@@ -430,6 +433,7 @@ _SoundPlayNode::LateNoticeReceived(const media_source& what, bigtime_t how_much,
 
 	// If we're late, we need to catch up.  Respond in a manner appropriate to our
 	// current run mode.
+/*
 	if (RunMode() == B_RECORDING)
 	{
 		// A hardware capture node can't adjust; it simply emits buffers at
@@ -438,13 +442,16 @@ _SoundPlayNode::LateNoticeReceived(const media_source& what, bigtime_t how_much,
 		// can't choose to capture "sooner"....
 	}
 	else if (RunMode() == B_INCREASE_LATENCY)
+*/
+	if (RunMode() != B_DROP_DATA)
 	{
 		// We're late, and our run mode dictates that we try to produce buffers
 		// earlier in order to catch up.  This argues that the downstream nodes are
 		// not properly reporting their latency, but there's not much we can do about
 		// that at the moment, so we try to start producing buffers earlier to
 		// compensate.
-		mInternalLatency += how_much;
+//		mInternalLatency += how_much;
+		mLatency += 1000;
 		SetEventLatency(mLatency + mInternalLatency);
 
 		fprintf(stderr, "\tincreasing latency to %Ld\n", mLatency + mInternalLatency);
@@ -552,39 +559,70 @@ status_t
 _SoundPlayNode::HandleBuffer(
 				const media_timed_event *event,
 				bigtime_t lateness,
-				bool realTimeEvent = false)
+				bool realTimeEvent)
 {
 	CALLED();
 		
 	// make sure we're both started *and* connected before delivering a buffer
-	if ((RunState() == BMediaEventLooper::B_STARTED) && (mOutput.destination != media_destination::null))
-	{
+	if ((RunState() != BMediaEventLooper::B_STARTED) || (mOutput.destination == media_destination::null))
+		return B_OK;
+		
+	// The event->event_time is the time at which the buffer we are preparing here should
+	// arrive at it's destination. The MediaEventLooper should have scheduled us early enough
+	// (based on EventLatency() and the SchedulingLatency()) to make this possible.
+
+	bigtime_t scheduling_latency = SchedulingLatency();
+
+	if (lateness > 0) {
+		printf("_SoundPlayNode::HandleBuffer, event sheduled too late, lateness is %Ld\n", lateness);
+		mInternalLatency += 1000;
+		SetEventLatency(mLatency + mInternalLatency);
+	}
+
+	// skip buffer creation if output not enabled	
+	if (mOutputEnabled) {
+
 		// Get the next buffer of data
 		BBuffer* buffer = FillNextBuffer(event->event_time);
-		if (buffer)
-		{
+
+		if (buffer) {
+
+			// If we are ready way too early, decrase internal latency
+			bigtime_t how_early = event->event_time - TimeSource()->Now() - mLatency;
+			if (how_early > (3 * scheduling_latency)) {
+
+				printf("_SoundPlayNode::HandleBuffer, event scheduled too early, how_early is %Ld\n", how_early);
+
+				if (mTooEarlyCount++ == 5) {
+					mInternalLatency -= how_early;
+					if (mInternalLatency < 500)
+						mInternalLatency = 500;
+					printf("_SoundPlayNode::HandleBuffer setting internal latency to %Ld\n", mInternalLatency);
+					SetEventLatency(mLatency + mInternalLatency);
+					mTooEarlyCount = 0;
+				}
+			}
+
 			// send the buffer downstream if and only if output is enabled
-			status_t err = B_ERROR;
-			if (mOutputEnabled) err = SendBuffer(buffer, mOutput.destination);
-			if (err)
-			{
-				// we need to recycle the buffer ourselves if output is disabled or
+			if (B_OK != SendBuffer(buffer, mOutput.destination)) {
+				// we need to recycle the buffer
 				// if the call to SendBuffer() fails
 				buffer->Recycle();
 			}
 		}
-
-		// track how much media we've delivered so far
-		size_t nFrames = mOutput.format.u.raw_audio.buffer_size
-			/ (mOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
-			/ mOutput.format.u.raw_audio.channel_count;
-		mFramesSent += nFrames;
-
-		// The buffer is on its way; now schedule the next one to go
-		bigtime_t nextEvent = mStartTime + bigtime_t(double(mFramesSent) / double(mOutput.format.u.raw_audio.frame_rate) * 1000000.0);
-		media_timed_event nextBufferEvent(nextEvent, BTimedEventQueue::B_HANDLE_BUFFER);
-		EventQueue()->AddEvent(nextBufferEvent);
 	}
+
+	// track how much media we've delivered so far
+	size_t nFrames = mOutput.format.u.raw_audio.buffer_size
+		/ (mOutput.format.u.raw_audio.format & media_raw_audio_format::B_AUDIO_SIZE_MASK)
+		/ mOutput.format.u.raw_audio.channel_count;
+	mFramesSent += nFrames;
+
+	// The buffer is on its way; now schedule the next one to go
+	// nextEvent is the time at which the buffer should arrive at it's destination
+	bigtime_t nextEvent = mStartTime + bigtime_t(double(mFramesSent) / double(mOutput.format.u.raw_audio.frame_rate) * 1000000.0);
+	media_timed_event nextBufferEvent(nextEvent, BTimedEventQueue::B_HANDLE_BUFFER);
+	EventQueue()->AddEvent(nextBufferEvent);
 	
 	return B_OK;
 }
@@ -690,6 +728,10 @@ _SoundPlayNode::AllocateBuffers()
 
 	DPRINTF("\tlatency = %Ld, buffer duration = %Ld\n", mLatency, BufferDuration());
 	DPRINTF("\tcreating group of %ld buffers, size = %lu\n", count, size);
+	
+	if (count < 2)
+		count == 2;
+	
 	mBufferGroup = new BBufferGroup(size, count);
 }
 
@@ -719,26 +761,8 @@ _SoundPlayNode::FillNextBuffer(bigtime_t event_time)
 	hdr->type = B_MEDIA_RAW_AUDIO;
 	hdr->size_used = mOutput.format.u.raw_audio.buffer_size;
 	hdr->time_source = TimeSource()->ID();
+	hdr->start_time = event_time;
 
-	bigtime_t stamp;
-	if (RunMode() == B_RECORDING)
-	{
-		// In B_RECORDING mode, we stamp with the capture time.  We're not
-		// really a hardware capture node, but we simulate it by using the (precalculated)
-		// time at which this buffer "should" have been created.
-		stamp = event_time;
-	}
-	else
-	{
-		// okay, we're in one of the "live" performance run modes.  in these modes, we
-		// stamp the buffer with the time at which the buffer should be rendered to the
-		// output, not with the capture time.  mStartTime is the cached value of the
-		// first buffer's performance time; we calculate this buffer's performance time as
-		// an offset from that time, based on the amount of media we've created so far.
-		// Recalculating every buffer like this avoids accumulation of error.
-		stamp = mStartTime + bigtime_t(double(mFramesSent) / double(mOutput.format.u.raw_audio.frame_rate) * 1000000.0);
-	}
-	hdr->start_time = stamp;
 	DPRINTF("TimeSource()->Now() : %li\n", TimeSource()->Now());
 	DPRINTF("hdr->start_time : %li\n", hdr->start_time);
 	DPRINTF("mFramesSent : %li\n", mFramesSent);
diff --git a/src/kits/media/SoundPlayNode.h b/src/kits/media/SoundPlayNode.h
index 597ef004e3..110c94681c 100644
--- a/src/kits/media/SoundPlayNode.h
+++ b/src/kits/media/SoundPlayNode.h
@@ -169,6 +169,7 @@ private:
 	bigtime_t 			mInternalLatency;
 	bigtime_t 			mStartTime;
 	uint64 				mFramesSent;
+	int32				mTooEarlyCount;
 };
 
 #endif
diff --git a/src/kits/media/SoundPlayer.cpp b/src/kits/media/SoundPlayer.cpp
index bfd6f27306..636e338581 100644
--- a/src/kits/media/SoundPlayer.cpp
+++ b/src/kits/media/SoundPlayer.cpp
@@ -155,11 +155,11 @@ BSoundPlayer::Start()
 	BTimeSource *timeSource = roster->MakeTimeSourceFor(_m_node->Node());
 	
 	// make sure we give the producer enough time to run buffers through
-			// the node chain, otherwise it'll start up already late
+	// the node chain, otherwise it'll start up already late
 	bigtime_t latency = 0;
 	status_t err = roster->GetLatencyFor(_m_node->Node(), &latency);
 
-	err = roster->StartNode(_m_node->Node(), timeSource->Now() + latency);
+	err = roster->StartNode(_m_node->Node(), timeSource->Now() + latency + 5000);
 	
 	timeSource->Release();
 	
diff --git a/src/kits/media/TimedEventQueuePrivate.cpp b/src/kits/media/TimedEventQueuePrivate.cpp
index f6cc9e4e4a..7de3cd621b 100644
--- a/src/kits/media/TimedEventQueuePrivate.cpp
+++ b/src/kits/media/TimedEventQueuePrivate.cpp
@@ -155,11 +155,13 @@ _event_queue_imp::RemoveFirstEvent(media_timed_event * outEvent)
 	if (fFirstEntry == 0)
 		return B_ERROR;
 	
-	if (outEvent != 0)
+	if (outEvent != 0) {
+		// No cleanup here
 		*outEvent = fFirstEntry->event;
-	else
+	} else {
 		CleanupEvent(&fFirstEntry->event);
-		
+	}
+
 	RemoveEntry(fFirstEntry);
 
 	return B_OK;
@@ -560,6 +562,7 @@ _event_queue_imp::CleanupEvent(media_timed_event *event)
 		// do nothing
 	} else if (event->type == BTimedEventQueue::B_HANDLE_BUFFER && event->cleanup == BTimedEventQueue::B_RECYCLE_BUFFER) {
 		((BBuffer *)event->pointer)->Recycle();
+		DEBUG_ONLY(*const_cast<void **>(&event->pointer) = NULL);
 	} else if (event->cleanup == BTimedEventQueue::B_EXPIRE_TIMER) {
 		// call TimerExpired() on the event->data
 		debugger("BTimedEventQueue cleanup: calling TimerExpired() should be implemented here\n");
diff --git a/src/servers/media/DefaultManager.cpp b/src/servers/media/DefaultManager.cpp
index 82e5e208dc..a6b5f102de 100644
--- a/src/servers/media/DefaultManager.cpp
+++ b/src/servers/media/DefaultManager.cpp
@@ -13,6 +13,9 @@
 /* no locking used in this file, we assume that the caller (NodeManager) does it.
  */
 
+
+#define MAX_NODE_INFOS 10
+
 DefaultManager::DefaultManager()
  :	fMixerConnected(false),
  	fPhysicalVideoOut(-1),
@@ -230,14 +233,14 @@ DefaultManager::FindPhysicalVideoIn()
 void
 DefaultManager::FindPhysicalAudioOut()
 {
-	live_node_info info[2];
+	live_node_info info[MAX_NODE_INFOS];
 	media_format input; /* a physical audio output has a logical data input */
 	int32 count;
 	status_t rv;
 
 	memset(&input, 0, sizeof(input));
 	input.type = B_MEDIA_RAW_AUDIO;
-	count = 2;
+	count = MAX_NODE_INFOS;
 	rv = BMediaRoster::Roster()->GetLiveNodes(&info[0], &count, &input, NULL, NULL, B_BUFFER_CONSUMER | B_PHYSICAL_OUTPUT);
 	if (rv != B_OK || count < 1) {
 		printf("Couldn't find physical audio output node\n");
@@ -249,7 +252,7 @@ DefaultManager::FindPhysicalAudioOut()
 	for (int i = 0; i < count; i++) {
 		if (0 == strcmp(info[i].name, "None Out")) // skip the Null audio driver
 			continue;
-		printf("Default physical audio output created!\n");
+		printf("Default physical audio output \"%s\" created!\n", info[i].name);
 		fPhysicalAudioOut = info[i].node.node;
 		return;
 	}
@@ -258,14 +261,14 @@ DefaultManager::FindPhysicalAudioOut()
 void
 DefaultManager::FindPhysicalAudioIn()
 {
-	live_node_info info[2];
+	live_node_info info[MAX_NODE_INFOS];
 	media_format output; /* a physical audio input has a logical data output */
 	int32 count;
 	status_t rv;
 
 	memset(&output, 0, sizeof(output));
 	output.type = B_MEDIA_RAW_AUDIO;
-	count = 2;
+	count = MAX_NODE_INFOS;
 	rv = BMediaRoster::Roster()->GetLiveNodes(&info[0], &count, NULL, &output, NULL, B_BUFFER_PRODUCER | B_PHYSICAL_INPUT);
 	if (rv != B_OK || count < 1) {
 		printf("Couldn't find physical audio input node\n");
@@ -277,7 +280,7 @@ DefaultManager::FindPhysicalAudioIn()
 	for (int i = 0; i < count; i++) {
 		if (0 == strcmp(info[i].name, "None In")) // skip the Null audio driver
 			continue;
-		printf("Default physical audio input created!\n");
+		printf("Default physical audio input \"%s\" created!\n", info[i].name);
 		fPhysicalAudioIn = info[i].node.node;
 		return;
 	}
@@ -286,26 +289,54 @@ DefaultManager::FindPhysicalAudioIn()
 void
 DefaultManager::FindTimeSource()
 {
-	live_node_info info;
+	live_node_info info[MAX_NODE_INFOS];
 	media_format input; /* a physical audio output has a logical data input (DAC)*/
 	int32 count;
 	status_t rv;
 	
-	/* We try to use the default physical audio out node,
-	 * as it most likely is a timesource.
-	 * XXX if that fails, we might use other audio or video clock timesources
+	/* First try to use the current default physical audio out
+	 */
+	if (fPhysicalAudioOut != -1) {
+		media_node clone;
+		if (B_OK == BMediaRoster::Roster()->GetNodeFor(fPhysicalAudioOut, &clone)) {
+			if (clone.kind & B_TIME_SOURCE) {
+				fTimeSource = clone.node;
+				BMediaRoster::Roster()->ReleaseNode(clone);
+				printf("Default DAC timesource created!\n");
+				return;
+			}
+			BMediaRoster::Roster()->ReleaseNode(clone);
+		} else {
+			printf("Default DAC is not a timesource!\n");
+		}
+	} else {
+		printf("Default DAC node does not exist!\n");
+	}
+	
+	/* Now try to find another physical audio out node
 	 */
-
 	memset(&input, 0, sizeof(input));
 	input.type = B_MEDIA_RAW_AUDIO;
-	count = 1;
-	rv = BMediaRoster::Roster()->GetLiveNodes(&info, &count, &input, NULL, NULL, B_TIME_SOURCE | B_PHYSICAL_OUTPUT);
-	if (rv != B_OK || count != 1) {
+	count = MAX_NODE_INFOS;
+	rv = BMediaRoster::Roster()->GetLiveNodes(&info[0], &count, &input, NULL, NULL, B_TIME_SOURCE | B_PHYSICAL_OUTPUT);
+	if (rv == B_OK && count >= 1) {
+		for (int i = 0; i < count; i++)
+			printf("info[%d].name %s\n", i, info[i].name);
+	
+		for (int i = 0; i < count; i++) {
+			// The BeOS R5 None Out node pretend to be a physical time source, that is pretty dumb
+			if (0 == strcmp(info[i].name, "None Out")) // skip the Null audio driver
+				continue;
+			printf("Default DAC timesource \"%s\" created!\n", info[i].name);
+			fTimeSource = info[i].node.node;
+			return;
+		}
+	} else {
 		printf("Couldn't find DAC timesource node\n");
-		return;
-	}
-	fTimeSource = info.node.node;
-	printf("Default DAC timesource created!\n");
+	}	
+	
+	/* XXX we might use other audio or video clock timesources
+	 */
 }
 
 void
@@ -334,6 +365,8 @@ DefaultManager::ConnectMixerToOutput()
 	media_node 			soundcard;
 	media_input 		input;
 	media_output 		output;
+	media_input 		newinput;
+	media_output 		newoutput;
 	media_format 		format;
 	BTimeSource * 		ts;
 	bigtime_t 			start_at;
@@ -372,20 +405,58 @@ DefaultManager::ConnectMixerToOutput()
 		rv = B_ERROR;
 		goto finish;
 	}
-
-	memset(&format, 0, sizeof(format));
-	format.type = B_MEDIA_RAW_AUDIO;
-	format.u.raw_audio.frame_rate = 44100;
-	format.u.raw_audio.channel_count = 2;
-	format.u.raw_audio.format = 0x2;
-
-	//roster->GetFormatFor(input, &format);
-
-	rv = roster->Connect(output.source, input.destination, &format, &output, &input);
-	if (rv != B_OK) {
-		printf("DefaultManager: connect failed\n");
-	}
 	
+	for (int i = 0; i < 5; i++) {
+		switch (i) {
+			case 0:
+				printf("DefaultManager: Trying connect in native format\n");
+				if (B_OK != roster->GetFormatFor(input, &format)) {
+					FATAL("DefaultManager: GetFormatFor failed\n");
+					continue;
+				}
+				break;
+			
+			case 1:
+				printf("DefaultManager: Trying connect in format 1\n");
+				memset(&format, 0, sizeof(format));
+				format.type = B_MEDIA_RAW_AUDIO;
+				format.u.raw_audio.frame_rate = 44100;
+				format.u.raw_audio.channel_count = 2;
+				format.u.raw_audio.format = 0x2;
+				break;
+
+			case 2:
+				printf("DefaultManager: Trying connect in format 2\n");
+				memset(&format, 0, sizeof(format));
+				format.type = B_MEDIA_RAW_AUDIO;
+				format.u.raw_audio.frame_rate = 48000;
+				format.u.raw_audio.channel_count = 2;
+				format.u.raw_audio.format = 0x2;
+				break;
+
+			case 3:
+				printf("DefaultManager: Trying connect in format 3\n");
+				memset(&format, 0, sizeof(format));
+				format.type = B_MEDIA_RAW_AUDIO;
+				break;
+
+			case 4:
+				printf("DefaultManager: Trying connect in format 4\n");
+				memset(&format, 0, sizeof(format));
+				break;
+		}
+		rv = roster->Connect(output.source, input.destination, &format, &newoutput, &newinput);
+		if (rv == B_OK)
+			break;
+	}
+	if (rv != B_OK) {
+		FATAL("DefaultManager: connect failed\n");
+		goto finish;
+	}
+
+	roster->SetRunModeNode(mixer, BMediaNode::B_INCREASE_LATENCY);
+	roster->SetRunModeNode(soundcard, BMediaNode::B_RECORDING);
+
 	roster->GetTimeSource(&timesource);
 	roster->SetTimeSourceFor(mixer.node, timesource.node);
 	roster->SetTimeSourceFor(soundcard.node, timesource.node);