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 d017a2d8ed..30135336ca 100644
--- a/src/add-ons/media/media-add-ons/mixer/AudioMixer.cpp
+++ b/src/add-ons/media/media-add-ons/mixer/AudioMixer.cpp
@@ -1,9 +1,11 @@
 /*
  * Copyright 2002 David Shipman,
  * Copyright 2003-2007 Marcus Overhagen
- * Copyright 2007 Haiku Inc. All rights reserved.
+ * Copyright 2007-2009 Haiku Inc. All rights reserved.
+ *
  * Distributed under the terms of the MIT License.
  */
+
 #include "AudioMixer.h"
 #include "MixerCore.h"
 #include "MixerInput.h"
@@ -31,1061 +33,6 @@
 
 #define USE_MEDIA_FORMAT_WORKAROUND 1
 
-#if USE_MEDIA_FORMAT_WORKAROUND
-static void 
-multi_audio_format_specialize(media_multi_audio_format *format,
-								const media_multi_audio_format *other);
-#endif
-
-#define FORMAT_USER_DATA_TYPE 		0x7294a8f3
-#define FORMAT_USER_DATA_MAGIC_1	0xc84173bd
-#define FORMAT_USER_DATA_MAGIC_2	0x4af62b7d
-
-AudioMixer::AudioMixer(BMediaAddOn *addOn, bool isSystemMixer)
-		:	BMediaNode("Audio Mixer"),
-			BBufferConsumer(B_MEDIA_RAW_AUDIO),
-			BBufferProducer(B_MEDIA_RAW_AUDIO),
-			BControllable(),
-			BMediaEventLooper(),
-			fAddOn(addOn),
-			fCore(new MixerCore(this)),
-			fWeb(0),
-			fBufferGroup(0),
-			fDownstreamLatency(1),
-			fInternalLatency(1),
-			fDisableStop(false)
-{
-	BMediaNode::AddNodeKind(B_SYSTEM_MIXER);
-
-	// this is the default format used for all wildcard format SpecializeTo()s
-	fDefaultFormat.type = B_MEDIA_RAW_AUDIO;
-	fDefaultFormat.u.raw_audio.frame_rate = 96000;
-	fDefaultFormat.u.raw_audio.channel_count = 2;
-	fDefaultFormat.u.raw_audio.format = media_raw_audio_format::B_AUDIO_FLOAT;
-	fDefaultFormat.u.raw_audio.byte_order = B_MEDIA_HOST_ENDIAN;
-	fDefaultFormat.u.raw_audio.buffer_size = 4096;
-	fDefaultFormat.u.raw_audio.channel_mask = 0;
-	fDefaultFormat.u.raw_audio.valid_bits = 0;
-	fDefaultFormat.u.raw_audio.matrix_mask = 0;
-	
-	if (isSystemMixer) {
-		// to get persistent settings, assign a settings file
-		BPath path;
-		if (B_OK != find_directory (B_USER_SETTINGS_DIRECTORY, &path))
-			path.SetTo("/boot/home/config/settings/");
-		path.Append("System Audio Mixer");
-		fCore->Settings()->SetSettingsFile(path.Path());
-		
-		// disable stop on the auto started (system) mixer
-		DisableNodeStop();
-	}
-	
-	ApplySettings();
-}
-
-AudioMixer::~AudioMixer()
-{
-	BMediaEventLooper::Quit();
-	SetParameterWeb(NULL); 
-
-	// stop the mixer
-	fCore->Lock();
-	fCore->Stop();
-	fCore->Unlock();
-
-	// disconnect all nodes from the mixer
-	// XXX todo
-	
-	delete fCore;
-	delete fBufferGroup;
-	
-	DEBUG_ONLY(fCore = 0; fBufferGroup = 0; fWeb = 0);
-}
-
-void
-AudioMixer::ApplySettings()
-{
-	fCore->Lock();
-	fCore->SetOutputAttenuation(fCore->Settings()->AttenuateOutput() ? 0.708 : 1.0);
-	fCore->Unlock();
-}
-
-void
-AudioMixer::DisableNodeStop()
-{
-	fDisableStop = true;
-}
-
-//
-// BMediaNode methods
-//
-
-void AudioMixer::Stop(bigtime_t performance_time, bool immediate)
-{
-	if (fDisableStop) {
-		TRACE("AudioMixer STOP is disabled\n");
-		return;
-	} else {
-		BMediaEventLooper::Stop(performance_time, immediate);
-	}
-}
-
-BMediaAddOn * 
-AudioMixer::AddOn(int32 *internal_id) const
-{
-	*internal_id = 0;
-	return fAddOn;
-}
-
-//
-// BBufferConsumer methods
-//
-
-status_t
-AudioMixer::HandleMessage(int32 message, const void *data, size_t size)
-{
-	// since we're using a mediaeventlooper, there shouldn't be any messages
-	return B_ERROR;
-}
-
-status_t
-AudioMixer::AcceptFormat(const media_destination &dest, media_format *ioFormat)
-{
-	PRINT_FORMAT("AudioMixer::AcceptFormat: ", *ioFormat);
-
-	// 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 have multiple inputs with different IDs, but
-	// the port number must match our ControlPort()
-	if (dest.port != ControlPort())
-		return B_MEDIA_BAD_DESTINATION;	
-
-	// specialize to raw audio format if necessary
-	if (ioFormat->type == B_MEDIA_UNKNOWN_TYPE)
-		ioFormat->type = B_MEDIA_RAW_AUDIO;
-	
-	// we require a raw audio format
-	if (ioFormat->type != B_MEDIA_RAW_AUDIO)
-		return B_MEDIA_BAD_FORMAT;
-	
-	// We do not have special requirements, but just in case
-	// another mixer is connecting to us and may need a hint
-	// to create a connection at optimal frame rate and
-	// channel count, we place this information in the user_data
-	fCore->Lock();
-	MixerOutput *output = fCore->Output();
-	uint32 channel_count = output ? output->MediaOutput().format.u.raw_audio.channel_count : 0;
-	float frame_rate = output ? output->MediaOutput().format.u.raw_audio.frame_rate : 0.0;
-	fCore->Unlock();
-	ioFormat->user_data_type = FORMAT_USER_DATA_TYPE;
-	*(uint32 *)&ioFormat->user_data[0] = FORMAT_USER_DATA_MAGIC_1;
-	*(uint32 *)&ioFormat->user_data[4] = channel_count;
-	*(float *)&ioFormat->user_data[20] = frame_rate;
-	*(uint32 *)&ioFormat->user_data[44] = FORMAT_USER_DATA_MAGIC_2;
-
-	return B_OK;
-}
-
-status_t
-AudioMixer::GetNextInput(int32 *cookie, media_input *out_input)
-{
-	TRACE("AudioMixer::GetNextInput\n");
-	
-	// our 0th input is always a wildcard and free one
-	if (*cookie == 0) {
-		out_input->node = Node();
-		out_input->source = media_source::null;
-		out_input->destination.port = ControlPort();
-		out_input->destination.id = 0;
-		memset(&out_input->format, 0, sizeof(out_input->format));
-		out_input->format.type = B_MEDIA_RAW_AUDIO;
-		strcpy(out_input->name, "Free Input");
-		*cookie += 1;
-		return B_OK;
-	}
-
-	// the other inputs are the currently connected ones
-	fCore->Lock();
-	MixerInput *input;
-	input = fCore->Input(*cookie - 1);
-	if (!input) {
-		fCore->Unlock();
-		return B_BAD_INDEX;
-	}
-	*out_input = input->MediaInput();
-	*cookie += 1;
-	fCore->Unlock();
-	return B_OK;
-}
-
-void
-AudioMixer::DisposeInputCookie(int32 cookie)
-{
-	// nothing to do
-}
-
-void
-AudioMixer::BufferReceived(BBuffer *buffer)
-{
-
-	if (buffer->Header()->type == B_MEDIA_PARAMETERS) {
-		TRACE("Control Buffer Received\n");
-		ApplyParameterData(buffer->Data(), buffer->SizeUsed());
-		buffer->Recycle();
-		return;
-	}
-
-	//PRINT(4, "buffer received at %12Ld, should arrive at %12Ld, delta %12Ld\n", TimeSource()->Now(), buffer->Header()->start_time, TimeSource()->Now() - buffer->Header()->start_time);
-
-	// Note: The following code is outcommented on purpose
-	// and is about to be modified at a later point
-	//	HandleInputBuffer(buffer, 0);
-	//	buffer->Recycle();
-	//	return;
-
-
-	// 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, bigtime_t lateness)
-{
-	// Note: The following code is outcommented on purpose
-	// and is about to be modified at a later point
-	/*
-	if (lateness > 5000) {
-		printf("Received buffer with way to high lateness %Ld\n", lateness);
-		if (RunMode() != B_DROP_DATA) {
-			printf("sending notify\n");
-			NotifyLateProducer(channel->fInput.source, lateness / 2, TimeSource()->Now());
-		} else if (RunMode() == B_DROP_DATA) {
-			printf("dropping buffer\n");
-			return;
-		}
-	}
-	*/
-
-	//	printf("Received buffer with lateness %Ld\n", lateness);
-	
-	fCore->Lock();
-	fCore->BufferReceived(buffer, lateness);
-	fCore->Unlock();
-
-	// Note: The following code is outcommented on purpose
-	// and is about to be modified at a later point
-	/*
-		if ((B_OFFLINE == RunMode()) && (B_DATA_AVAILABLE == channel->fProducerDataStatus))
-		{
-			RequestAdditionalBuffer(channel->fInput.source, buffer);
-		}
-	*/
-}
-
-void
-AudioMixer::ProducerDataStatus( const media_destination &for_whom, 
-								int32 status, bigtime_t at_performance_time)
-{
-/*	
-	if (IsValidDest(for_whom))
-	{
-		media_timed_event event(at_performance_time, BTimedEventQueue::B_DATA_STATUS,
-			(void *)(&for_whom), BTimedEventQueue::B_NO_CLEANUP, status, 0, NULL);
-		EventQueue()->AddEvent(event);
-		
-		// FIX_THIS
-		// the for_whom destination is not being sent correctly - verify in HandleEvent loop
-		
-	}
-*/
-}
-
-status_t
-AudioMixer::GetLatencyFor(const media_destination &for_whom,
-						  bigtime_t *out_latency,
-						  media_node_id *out_timesource)
-{
-	// we have multiple inputs with different IDs, but
-	// the port number must match our ControlPort()
-	if (for_whom.port != ControlPort())
-		return B_MEDIA_BAD_DESTINATION;	
-		
-	// return our event latency - this includes our internal + downstream 
-	// latency, but _not_ the scheduling latency
-	*out_latency = EventLatency();
-	*out_timesource = TimeSource()->ID();
-
-	printf("AudioMixer::GetLatencyFor %Ld, timesource is %ld\n", 
-							*out_latency,
-							*out_timesource);
-
-	return B_OK;
-}
-
-status_t
-AudioMixer::Connected(const media_source &producer, const media_destination &where,
-					  const media_format &with_format, media_input *out_input)
-{
-	PRINT_FORMAT("AudioMixer::Connected: ", with_format);
-
-	// workaround for a crashing bug in RealPlayer.  to be proper, RealPlayer's
-	// BBufferProducer::PrepareToConnect() should have removed all wildcards.
-	if (out_input->format.u.raw_audio.frame_rate == 0) {
-		fprintf(stderr, "Audio Mixer Warning: "
-					"Producer (port %ld, id %ld) connected with frame_rate=0\n",
-					producer.port,
-					producer.id);
-		MixerOutput *output = fCore->Output();
-		float frame_rate = output ? output->MediaOutput().format.u.raw_audio.frame_rate : 44100.0f;
-		out_input->format.u.raw_audio.frame_rate = frame_rate;
-	}
-	
-	// a BBufferProducer is connecting to our BBufferConsumer
-			
-	// incoming connections should always have an incoming ID=0,
-	// and the port number must match our ControlPort()
-	if (where.id != 0 || where.port != ControlPort())
-		return B_MEDIA_BAD_DESTINATION;
-	
-	fCore->Lock();
-	
-	// we assign a new id (!= 0) to the newly created input
-	out_input->destination.id = fCore->CreateInputID();
-
-	// 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 the input has no name, assign one
-	if (strlen(out_input->name) == 0)
-		sprintf(out_input->name, "Input %ld", out_input->destination.id);
-	
-	// add a new input to the mixer engine
-	MixerInput *input;
-	input = fCore->AddInput(*out_input);
-
-	fCore->Settings()->LoadConnectionSettings(input);
-
-	fCore->Unlock();
-	
-	// 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.
-	// But we have no special buffer requirements anyway...
-
-	UpdateParameterWeb();
-
-	return B_OK;
-}
-
-void
-AudioMixer::Disconnected(const media_source &producer, const media_destination &where)
-{
-	// One of our inputs has been disconnected
-
-	// check if it is really belongs to us
-	if (where.port != ControlPort()) {
-		TRACE("AudioMixer::Disconnected wrong input port\n");
-		return;
-	}
-	
-	fCore->Lock();
-	
-	if (!fCore->RemoveInput(where.id)) {
-		TRACE("AudioMixer::Disconnected can't remove input\n");
-	}
-	
-	fCore->Unlock();
-	UpdateParameterWeb();
-}
-
-status_t
-AudioMixer::FormatChanged(const media_source &producer, const media_destination &consumer, 
-						  int32 change_tag, const media_format &format)
-{
-	// at some point in the future (indicated by change_tag and RequestCompleted()),
-	// we will receive buffers in a different format
-
-	TRACE("AudioMixer::FormatChanged\n");
-
-	if (consumer.port != ControlPort() || consumer.id == 0)
-		return B_MEDIA_BAD_DESTINATION;
-
-	if (fCore->Settings()->RefuseInputFormatChange()) {
-		TRACE("AudioMixer::FormatChanged: input format change refused\n");
-		return B_ERROR;
-	}
-	
-	// XXX we should not apply the format change at this point
-
-	// tell core about format change
-	fCore->Lock();
-	fCore->InputFormatChanged(consumer.id, format.u.raw_audio);
-	fCore->Unlock();
-	
-	return B_OK;
-}
-
-//
-// BBufferProducer methods
-//
-
-status_t
-AudioMixer::FormatSuggestionRequested(media_type type, int32 quality, media_format *format)
-{
-	TRACE("AudioMixer::FormatSuggestionRequested\n");
-	
-	// BBufferProducer function, a downstream consumer
-	// is asking for our output format
-
-	if (type != B_MEDIA_RAW_AUDIO && type != B_MEDIA_UNKNOWN_TYPE)
-		return B_MEDIA_BAD_FORMAT;
-
-	// we can produce any (wildcard) raw audio format
-	memset(format, 0, sizeof(*format));
-	format->type = B_MEDIA_RAW_AUDIO;
-	return B_OK;
-}
-
-status_t
-AudioMixer::FormatProposal(const media_source &output, media_format *ioFormat)
-{
-	// BBufferProducer function, we implement this function to verify that the
-	// proposed media_format is suitable for the specified output. If any fields
-	// in the format are wildcards, and we have a specific requirement, adjust
-	// those fields to match our requirements before returning.
-
-	TRACE("AudioMixer::FormatProposal\n");
-	
-	// we only have one output (id=0, port=ControlPort())
-	if (output.id != 0 || output.port != ControlPort())
-		return B_MEDIA_BAD_SOURCE;
-
-	// specialize to raw audio format if necessary
-	if (ioFormat->type == B_MEDIA_UNKNOWN_TYPE)
-		ioFormat->type = B_MEDIA_RAW_AUDIO;
-	
-	// we require a raw audio format
-	if (ioFormat->type != B_MEDIA_RAW_AUDIO)
-		return B_MEDIA_BAD_FORMAT;
-
-	return B_OK;
-}
-
-//	If the format isn't good, put a good format into *io_format and return error
-//	If format has wildcard, specialize to what you can do (and change).
-//	If you can change the format, return OK.
-//	The request comes from your destination sychronously, so you cannot ask it
-//	whether it likes it -- you should assume it will since it asked.
-status_t
-AudioMixer::FormatChangeRequested(const media_source &source, const media_destination &destination, 
-								  media_format *io_format, int32 *_deprecated_)
-{
-	// the downstream consumer node (soundcard) requested that we produce
-	// another format, we need to check if the format is acceptable and
-	// remove any wildcards before returning OK.
-	
-	TRACE("AudioMixer::FormatChangeRequested\n");
-
-	if (fCore->Settings()->RefuseOutputFormatChange()) {
-		TRACE("AudioMixer::FormatChangeRequested: output format change refused\n");
-		return B_ERROR;
-	}
-
-	fCore->Lock();
-
-	MixerOutput *output = fCore->Output();
-	if (!output) {
-		ERROR("AudioMixer::FormatChangeRequested: no output\n");
-		goto err;
-	}
-	if (source != output->MediaOutput().source) {
-		ERROR("AudioMixer::FormatChangeRequested: wrong output source\n");
-		goto err;
-	}
-	if (destination != output->MediaOutput().destination) {
-		ERROR("AudioMixer::FormatChangeRequested: wrong output destination (port %ld, id %ld), our is (port %ld, id %ld)\n", destination.port, destination.id, output->MediaOutput().destination.port, output->MediaOutput().destination.id);
-		if (destination.port == output->MediaOutput().destination.port && destination.id == output->MediaOutput().destination.id + 1) {
-			ERROR("AudioMixer::FormatChangeRequested: this might be the broken R5 multi audio add-on\n");
-			goto err;
-		} else {
-			goto err;
-		}
-	}
-	if (io_format->type != B_MEDIA_RAW_AUDIO && io_format->type != B_MEDIA_UNKNOWN_TYPE) {
-		ERROR("AudioMixer::FormatChangeRequested: wrong format type\n");
-		goto err;
-	}
-	
-	/* remove wildcards */
-	#if USE_MEDIA_FORMAT_WORKAROUND
-		multi_audio_format_specialize(&io_format->u.raw_audio, &fDefaultFormat.u.raw_audio);
-	#else 
-		io_format->SpecializeTo(&fDefaultFormat);
-	#endif
-	
-	media_node_id id;
-	FindLatencyFor(destination, &fDownstreamLatency, &id);
-	TRACE("AudioMixer: Downstream Latency is %Ld usecs\n", fDownstreamLatency);
-
-	// SetDuration of one buffer
-	SetBufferDuration(buffer_duration(io_format->u.raw_audio));
-	TRACE("AudioMixer: buffer duration is %Ld usecs\n", BufferDuration());
-
-	// Our internal latency is at least the length of a full output buffer
-	fInternalLatency = BufferDuration() + max(4500LL, bigtime_t(0.5 * BufferDuration()));
-	TRACE("AudioMixer: Internal latency is %Ld usecs\n", fInternalLatency);
-	
-	SetEventLatency(fDownstreamLatency + fInternalLatency);
-	
-	// we need to inform all connected *inputs* about *our* change in latency
-	PublishEventLatencyChange();
-	
-	// XXX we might need to create more buffers, to span a larger downstream latency
-
-	// apply latency change
-	fCore->SetTimingInfo(TimeSource(), fDownstreamLatency);
-
-	// apply format change
-	fCore->OutputFormatChanged(io_format->u.raw_audio);
-
-	delete fBufferGroup;
-	fBufferGroup = CreateBufferGroup();
-	fCore->SetOutputBufferGroup(fBufferGroup);
-
-	fCore->Unlock();
-	return B_OK;
-	
-err:
-	fCore->Unlock();
-	return B_ERROR;
-}
-
-status_t 
-AudioMixer::GetNextOutput(int32 *cookie, media_output *out_output)
-{
-	TRACE("AudioMixer::GetNextOutput\n");
-
-	if (*cookie != 0)
-		return B_BAD_INDEX;
-
-	fCore->Lock();
-	MixerOutput *output = fCore->Output();
-	if (output) {
-		*out_output = output->MediaOutput();
-	} else {
-		out_output->node = Node();
-		out_output->source.port = ControlPort();
-		out_output->source.id = 0;
-		out_output->destination = media_destination::null;
-		memset(&out_output->format, 0, sizeof(out_output->format));
-		out_output->format.type = B_MEDIA_RAW_AUDIO;
-		strcpy(out_output->name, "Mixer Output");
-	}
-	fCore->Unlock();
-
-	*cookie += 1;
-	return B_OK;
-}
-
-status_t 
-AudioMixer::DisposeOutputCookie(int32 cookie)
-{
-	// nothing to do
-	return B_OK;
-}
-
-status_t 
-AudioMixer::SetBufferGroup(const media_source &for_source, BBufferGroup *newGroup)
-{
-	PRINT("AudioMixer::SetBufferGroup\n");
-	// the downstream consumer (soundcard) node asks us to use another
-	// BBufferGroup (might be NULL). We only have one output (id 0)
-	if (for_source.port != ControlPort() || for_source.id != 0)
-		return B_MEDIA_BAD_SOURCE;
-
-	if (newGroup == fBufferGroup) {
-		// we're already using this buffergroup
-		return B_OK;
-	}
-	
-	fCore->Lock();
-	if (!newGroup)
-		newGroup = CreateBufferGroup();
-	fCore->SetOutputBufferGroup(newGroup);
-	delete fBufferGroup;
-	fBufferGroup = newGroup;
-	fCore->Unlock();
-
-	return B_OK;
-}
-
-status_t
-AudioMixer::GetLatency(bigtime_t *out_latency)
-{
-	// report our *total* latency:  internal plus downstream plus scheduling
-	*out_latency = EventLatency() + SchedulingLatency();
-
-	TRACE("AudioMixer::GetLatency %Ld\n", *out_latency);
-
-	return B_OK;
-}
-
-void
-AudioMixer::LatencyChanged(const media_source & source, const media_destination & destination,
-						   bigtime_t new_latency, uint32 flags)
-{
-	if (source.port != ControlPort() || source.id != 0) {
-		ERROR("AudioMixer::LatencyChanged: received but has wrong source %ld/%ld\n", source.port, source.id);
-		return;
-	}
-
-	TRACE("AudioMixer::LatencyChanged: downstream latency from %ld/%ld to %ld/%ld changed from %Ld to %Ld\n",
-		source.port, source.id, destination.port, destination.id, fDownstreamLatency, new_latency);
-
-#if DEBUG
-	{
-		media_node_id id;
-		bigtime_t l;
-		FindLatencyFor(destination, &l, &id);
-		TRACE("AudioMixer: Reported downstream Latency is %Ld usecs\n", l);
-	}
-#endif
-
-	fDownstreamLatency = new_latency;
-	SetEventLatency(fDownstreamLatency + fInternalLatency);
-
-	// XXX we might need to create more buffers, to span a larger downstream latency
-	
-	fCore->Lock();
-	fCore->SetTimingInfo(TimeSource(), fDownstreamLatency);
-	PublishEventLatencyChange();
-	fCore->Unlock();
-}
-
-status_t
-AudioMixer::PrepareToConnect(const media_source &what, 
-								const media_destination &where,
-								media_format *format, 
-								media_source *out_source,
-								char *out_name)
-{
-	TRACE("AudioMixer::PrepareToConnect\n");
-	// PrepareToConnect() is the second stage of format negotiations that happens
-	// inside BMediaRoster::Connect(). At this point, the consumer's AcceptFormat()
-	// method has been called, and that node has potentially changed the proposed
-	// format. It may also have left wildcards in the format. PrepareToConnect()
-	// *must* fully specialize the format before returning!
-	// we also create the new output connection and return it in out_source.
-
-	PRINT_FORMAT("AudioMixer::PrepareToConnect: suggested format", *format);
-
-	// avoid loop connections
-	if (where.port == ControlPort())
-		return B_MEDIA_BAD_SOURCE;
-
-	// is the source valid?
-	if (what.port != ControlPort() || what.id != 0)
-		return B_MEDIA_BAD_SOURCE;
-
-	// is the format acceptable?
-	if (format->type != B_MEDIA_RAW_AUDIO && format->type != B_MEDIA_UNKNOWN_TYPE) {
-		PRINT_FORMAT("AudioMixer::PrepareToConnect: bad format", *format);
-		return B_MEDIA_BAD_FORMAT;
-	}
-
-	fCore->Lock();
-
-	// are we already connected?
-	if (fCore->Output() != 0) {
-		fCore->Unlock();
-		ERROR("AudioMixer::PrepareToConnect: already connected\n");
-		return B_MEDIA_ALREADY_CONNECTED;
-	}
-
-	// It is possible that another mixer is connecting.
-	// To avoid using the default format, we use one of
-	// a) the format that it indicated as hint in the user_data,
-	// b) the output format of the system audio mixer
-	// c) the input format of the system DAC device
-	// d) if everything failes, keep the wildcard
-	if (format->u.raw_audio.channel_count == 0
-		&& format->u.raw_audio.frame_rate < 1
-		&& format->user_data_type == FORMAT_USER_DATA_TYPE
-		&& *(uint32 *)&format->user_data[0] == FORMAT_USER_DATA_MAGIC_1
-		&& *(uint32 *)&format->user_data[44] == FORMAT_USER_DATA_MAGIC_2) {
-		// ok, a mixer is connecting
-		uint32 channel_count = *(uint32 *)&format->user_data[4];
-		float frame_rate = *(float *)&format->user_data[20];
-		if (channel_count > 0 && frame_rate > 0) {
-			// format is good, use it
-			format->u.raw_audio.channel_count = channel_count;
-			format->u.raw_audio.frame_rate = frame_rate;
-		} else {
-			// other mixer's output is probably not connected
-			media_node node;
-			BMediaRoster *roster = BMediaRoster::Roster();
-			media_output out;
-			media_input in;
-			int32 count;
-			if (B_OK == roster->GetAudioMixer(&node) && B_OK == roster->GetConnectedOutputsFor(node, &out, 1, &count) && count == 1) {
-				// use mixer output format
-				format->u.raw_audio.channel_count = out.format.u.raw_audio.channel_count;
-				format->u.raw_audio.frame_rate = out.format.u.raw_audio.frame_rate;
-			} else if (B_OK == roster->GetAudioOutput(&node) && B_OK == roster->GetAllInputsFor(node, &in, 1, &count) && count == 1) {
-				// use DAC input format
-				format->u.raw_audio.channel_count = in.format.u.raw_audio.channel_count;
-				format->u.raw_audio.frame_rate = in.format.u.raw_audio.frame_rate;
-			}
-		}
-	}
-
-	/* set source and suggest a name */
-	*out_source = what;
-	strcpy(out_name, "Mixer Output");
-
-	/* remove wildcards */
-	#if USE_MEDIA_FORMAT_WORKAROUND
-		multi_audio_format_specialize(&format->u.raw_audio, &fDefaultFormat.u.raw_audio);
-	#else 
-		format->SpecializeTo(&fDefaultFormat);
-	#endif
-
-	PRINT_FORMAT("AudioMixer::PrepareToConnect: final format", *format);
-
-	/* add output to core */	
-	media_output output;
-	output.node = Node();
-	output.source = *out_source;
-	output.destination = where;
-	output.format = *format;
-	strcpy(output.name, out_name);
-	
-	fCore->EnableOutput(false);
-	fCore->AddOutput(output);
-
-	fCore->Unlock();
-	return B_OK;
-}
-
-void 
-AudioMixer::Connect(status_t error, const media_source &source, 
-					const media_destination &dest, 
-					const media_format &format,
-					char *io_name)
-{
-	TRACE("AudioMixer::Connect\n");
-
-	fCore->Lock();
-	// are we still connected?
-	if (fCore->Output() == 0) {
-		fCore->Unlock();
-		ERROR("AudioMixer::Connect: no longer connected\n");
-		return;
-	}
-	fCore->Unlock();
-
-	if (error != B_OK) {
-		// if an error occured, remove output from core
-		ERROR("AudioMixer::Connect failed with error 0x%08lX, removing connection\n", error);
-		fCore->Lock();
-		fCore->RemoveOutput();
-		fCore->Unlock();
-		return;
-	}
-
-	/* Switch our prefered format to have the same
-	 * frame_rate and channel count as the output.
-	 */
-	fDefaultFormat.u.raw_audio.frame_rate = format.u.raw_audio.frame_rate;
-	fDefaultFormat.u.raw_audio.channel_count = format.u.raw_audio.channel_count;
-
-	// if the connection has no name, we set it now
-	if (strlen(io_name) == 0)
-		strcpy(io_name, "Mixer Output");
-		
-	// Now that we're connected, we can determine our downstream latency.
-	media_node_id id;
-	FindLatencyFor(dest, &fDownstreamLatency, &id);
-	TRACE("AudioMixer: Downstream Latency is %Ld usecs\n", fDownstreamLatency);
-
-	// SetDuration of one buffer
-	SetBufferDuration(buffer_duration(format.u.raw_audio));
-	TRACE("AudioMixer: buffer duration is %Ld usecs\n", BufferDuration());
-
-	// Our internal latency is at least the length of a full output buffer
-	fInternalLatency = BufferDuration() + max(4500LL, bigtime_t(0.5 * BufferDuration()));
-	TRACE("AudioMixer: Internal latency is %Ld usecs\n", fInternalLatency);
-	
-	SetEventLatency(fDownstreamLatency + fInternalLatency);
-	
-	// we need to inform all connected *inputs* about *our* change in latency
-	PublishEventLatencyChange();
-
-	// 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)
-		fBufferGroup = CreateBufferGroup();
-
-	fCore->Lock();
-
-	ASSERT(fCore->Output() != 0);
-	
-	// our source should still be valid, too
-	ASSERT(fCore->Output()->MediaOutput().source.id == 0);
-	ASSERT(fCore->Output()->MediaOutput().source.port == ControlPort());
-	
-	// BBufferConsumer::Connected() may return a different input for the
-	// newly created connection. The destination can have changed since
-	// AudioMixer::PrepareToConnect() and we need to update it.
-	fCore->Output()->MediaOutput().destination = dest;
-
-	fCore->EnableOutput(true);
-	fCore->SetTimingInfo(TimeSource(), fDownstreamLatency);
-	fCore->SetOutputBufferGroup(fBufferGroup);
-	
-	fCore->Settings()->LoadConnectionSettings(fCore->Output());
-
-	fCore->Unlock();
-	UpdateParameterWeb();
-}
-
-void 
-AudioMixer::Disconnect(const media_source &what, const media_destination &where)
-{
-	TRACE("AudioMixer::Disconnect\n");
-	fCore->Lock();
-
-	// Make sure that our connection is the one being disconnected
-	MixerOutput * output = fCore->Output();
-	if (!output || output->MediaOutput().node != Node() || output->MediaOutput().source != what || output->MediaOutput().destination != where) {
-		ERROR("AudioMixer::Disconnect can't disconnect (wrong connection)\n");
-		fCore->Unlock();
-		return;
-	}
-
-	// Switch our prefered format back to default
-	// frame rate and channel count.
-	fDefaultFormat.u.raw_audio.frame_rate = 96000;
-	fDefaultFormat.u.raw_audio.channel_count = 2;
-	
-	// force a stop
-	fCore->Stop();
-	
-	fCore->RemoveOutput();
-
-	// destroy buffer group
-	delete fBufferGroup;
-	fBufferGroup = 0;
-	fCore->SetOutputBufferGroup(0);
-	
-	fCore->Unlock();
-	UpdateParameterWeb();
-}
-
-
-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
-
-	ERROR("AudioMixer::LateNoticeReceived, %Ld too late at %Ld\n", how_much, performance_time);
-	// Note: The following code is outcommented on purpose
-	// and is about to be modified at a later point
-	/*	
-	if (what == fOutput.source) {
-		if (RunMode() == B_INCREASE_LATENCY) {
-			fInternalLatency += how_much;
-
-			if (fInternalLatency > 50000)
-				fInternalLatency = 50000;
-
-			printf("AudioMixer: increasing internal latency to %Ld usec\n", fInternalLatency);
-			SetEventLatency(fDownstreamLatency + fInternalLatency);
-			
-			PublishEventLatencyChange();
-		}
-	}
-	*/
-}
-
-
-void 
-AudioMixer::EnableOutput(const media_source &what, bool enabled, int32 *_deprecated_)
-{
-	// we only have one output
-	if (what.id != 0 || what.port != ControlPort())
-		return;
-		
-	fCore->Lock();
-	fCore->EnableOutput(enabled);
-	fCore->Unlock();
-}
-
-
-//
-// BMediaEventLooper methods
-//
-
-void
-AudioMixer::NodeRegistered()
-{
-	Run();
-	SetPriority(120);
-	UpdateParameterWeb();
-}
-
-void
-AudioMixer::SetTimeSource(BTimeSource * time_source)
-{
-	TRACE("AudioMixer::SetTimeSource: timesource is now %ld\n", time_source->ID());
-	fCore->Lock();
-	fCore->SetTimingInfo(time_source, fDownstreamLatency);
-	fCore->Unlock();
-}
-
-void 
-AudioMixer::HandleEvent(const media_timed_event *event, bigtime_t lateness, bool realTimeEvent)
-{
-	switch (event->type)
-	{
-	
-		case BTimedEventQueue::B_HANDLE_BUFFER:
-		{
-			HandleInputBuffer((BBuffer *)event->pointer, lateness);
-			((BBuffer *)event->pointer)->Recycle();
-			break;
-		}
-
-		case BTimedEventQueue::B_START:
-		{
-			TRACE("AudioMixer::HandleEvent: B_START\n");
-			if (RunState() != B_STARTED) {
-				fCore->Lock();
-				fCore->Start();
-				fCore->Unlock();
-			}
-			break;
-		}
-			
-		case BTimedEventQueue::B_STOP:
-		{
-			TRACE("AudioMixer::HandleEvent: B_STOP\n");
-			// stopped - don't process any more buffers, flush all buffers from eventqueue
-			EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true, BTimedEventQueue::B_HANDLE_BUFFER);
-			fCore->Lock();
-			fCore->Stop();
-			fCore->Unlock();
-			break;
-		}
-
-		case BTimedEventQueue::B_DATA_STATUS:
-		{
-			ERROR("DataStatus message\n");
-			break;
-		}
-		
-		default:
-			break;	
-	
-	}
-}
-
-//
-// AudioMixer methods
-//		
-
-void
-AudioMixer::PublishEventLatencyChange()
-{
-	// our event (processing + downstream) latency has changed,
-	// and we need tell all inputs about this
-	
-	TRACE("AudioMixer::PublishEventLatencyChange\n");
-
-	fCore->Lock();
-
-	MixerInput *input;	
-	for (int i = 0; (input = fCore->Input(i)) != 0; i++) {
-		TRACE("AudioMixer::PublishEventLatencyChange: SendLatencyChange, connection %ld/%ld to %ld/%ld event latency is now %Ld\n",
-			input->MediaInput().source.port, input->MediaInput().source.id,
-			input->MediaInput().destination.port, input->MediaInput().destination.id,
-			EventLatency());
-		SendLatencyChange(input->MediaInput().source, input->MediaInput().destination, EventLatency());
-	}
-
-	fCore->Unlock();
-}
-
-BBufferGroup *
-AudioMixer::CreateBufferGroup()
-{
-	// allocate enough buffers to span our downstream latency 
-	// (plus one for rounding up), plus one extra
-	int32 count = int32(fDownstreamLatency / BufferDuration()) + 2;
-
-	TRACE("AudioMixer::CreateBufferGroup: fDownstreamLatency %Ld,  BufferDuration %Ld, buffer count = %ld\n", fDownstreamLatency, BufferDuration(), count);
-
-	if (count < 3)
-		count = 3;
-
-	fCore->Lock();
-	uint32 size = fCore->Output()->MediaOutput().format.u.raw_audio.buffer_size;
-	fCore->Unlock();
-
-	TRACE("AudioMixer: allocating %ld buffers of %ld bytes each\n", count, size);
-	return new BBufferGroup(size, count);
-}
-
-float
-AudioMixer::dB_to_Gain(float db)
-{
-	TRACE("dB_to_Gain: dB in: %01.2f ", db);
-	if (fCore->Settings()->NonLinearGainSlider()) {
-		if (db > 0) {
-			db = db * (pow(abs(DB_MAX), (1.0 / DB_EXPONENT_POSITIVE)) / abs(DB_MAX));
-			db = pow(db, DB_EXPONENT_POSITIVE);
-		} else {
-			db = -db;
-			db = db * (pow(abs(DB_MIN), (1.0 / DB_EXPONENT_NEGATIVE)) / abs(DB_MIN));
-			db = pow(db, DB_EXPONENT_NEGATIVE);
-			db = -db;
-		}
-	}
-	TRACE("dB out: %01.2f\n", db);
-	return pow(10.0, db / 20.0);
-}
-
-float
-AudioMixer::Gain_to_dB(float gain)
-{
-	float db;
-	db = 20.0 * log10(gain);
-	if (fCore->Settings()->NonLinearGainSlider()) {
-		if (db > 0) {
-			db = pow(db, (1.0 / DB_EXPONENT_POSITIVE));
-			db = db * (abs(DB_MAX) / pow(abs(DB_MAX), (1.0 / DB_EXPONENT_POSITIVE)));
-		} else {
-			db = -db;
-			db = pow(db, (1.0 / DB_EXPONENT_NEGATIVE));
-			db = db * (abs(DB_MIN) / pow(abs(DB_MIN), (1.0 / DB_EXPONENT_NEGATIVE)));
-			db = -db;
-		}
-	}
-	return db;
-}
-
-//
-// BControllable methods
-//
-
 #define DB_TO_GAIN(db)			dB_to_Gain((db))
 #define GAIN_TO_DB(gain)		Gain_to_dB((gain))
 #define PERCENT_TO_GAIN(pct)	((pct) / 100.0)
@@ -1125,8 +72,1130 @@ AudioMixer::Gain_to_dB(float gain)
 #define PARAM_IS_GAIN(id)					(((id) & 0xf) == 0x8)
 #define PARAM_IS_BALANCE(id)				(((id) & 0xf) == 0x9)
 
+#if USE_MEDIA_FORMAT_WORKAROUND
+static void
+multi_audio_format_specialize(media_multi_audio_format *format,
+	const media_multi_audio_format *other);
+#endif
 
-status_t 
+#define FORMAT_USER_DATA_TYPE 		0x7294a8f3
+#define FORMAT_USER_DATA_MAGIC_1	0xc84173bd
+#define FORMAT_USER_DATA_MAGIC_2	0x4af62b7d
+
+
+AudioMixer::AudioMixer(BMediaAddOn *addOn, bool isSystemMixer)
+	: BMediaNode("Audio Mixer"),
+	BBufferConsumer(B_MEDIA_RAW_AUDIO),
+	BBufferProducer(B_MEDIA_RAW_AUDIO),
+	BControllable(),
+	BMediaEventLooper(),
+	fAddOn(addOn),
+	fCore(new MixerCore(this)),
+	fWeb(0),
+	fBufferGroup(0),
+	fDownstreamLatency(1),
+	fInternalLatency(1),
+	fDisableStop(false)
+{
+	BMediaNode::AddNodeKind(B_SYSTEM_MIXER);
+
+	// this is the default format used for all wildcard format SpecializeTo()s
+	fDefaultFormat.type = B_MEDIA_RAW_AUDIO;
+	fDefaultFormat.u.raw_audio.frame_rate = 96000;
+	fDefaultFormat.u.raw_audio.channel_count = 2;
+	fDefaultFormat.u.raw_audio.format = media_raw_audio_format::B_AUDIO_FLOAT;
+	fDefaultFormat.u.raw_audio.byte_order = B_MEDIA_HOST_ENDIAN;
+	fDefaultFormat.u.raw_audio.buffer_size = 4096;
+	fDefaultFormat.u.raw_audio.channel_mask = 0;
+	fDefaultFormat.u.raw_audio.valid_bits = 0;
+	fDefaultFormat.u.raw_audio.matrix_mask = 0;
+
+	if (isSystemMixer) {
+		// to get persistent settings, assign a settings file
+		BPath path;
+		if (B_OK != find_directory (B_USER_SETTINGS_DIRECTORY, &path))
+			path.SetTo("/boot/home/config/settings/");
+		path.Append("System Audio Mixer");
+		fCore->Settings()->SetSettingsFile(path.Path());
+
+		// disable stop on the auto started (system) mixer
+		DisableNodeStop();
+	}
+
+	ApplySettings();
+}
+
+
+AudioMixer::~AudioMixer()
+{
+	BMediaEventLooper::Quit();
+	SetParameterWeb(NULL);
+
+	// stop the mixer
+	fCore->Lock();
+	fCore->Stop();
+	fCore->Unlock();
+
+	// disconnect all nodes from the mixer
+	// XXX todo
+
+	delete fCore;
+	delete fBufferGroup;
+
+	DEBUG_ONLY(fCore = 0; fBufferGroup = 0; fWeb = 0);
+}
+
+
+void
+AudioMixer::ApplySettings()
+{
+	fCore->Lock();
+	fCore->SetOutputAttenuation(fCore->Settings()->AttenuateOutput() ? 0.708 : 1.0);
+	fCore->Unlock();
+}
+
+
+void
+AudioMixer::DisableNodeStop()
+{
+	fDisableStop = true;
+}
+
+
+//	#pragma mark - BMediaNode methods
+
+
+void
+AudioMixer::Stop(bigtime_t performance_time, bool immediate)
+{
+	if (fDisableStop) {
+		TRACE("AudioMixer STOP is disabled\n");
+		return;
+	} else {
+		BMediaEventLooper::Stop(performance_time, immediate);
+	}
+}
+
+
+BMediaAddOn*
+AudioMixer::AddOn(int32 *internal_id) const
+{
+	*internal_id = 0;
+	return fAddOn;
+}
+
+
+//	#pragma mark - BBufferConsumer methods
+
+
+status_t
+AudioMixer::HandleMessage(int32 message, const void *data, size_t size)
+{
+	// since we're using a mediaeventlooper, there shouldn't be any messages
+	return B_ERROR;
+}
+
+
+status_t
+AudioMixer::AcceptFormat(const media_destination &dest, media_format *ioFormat)
+{
+	PRINT_FORMAT("AudioMixer::AcceptFormat: ", *ioFormat);
+
+	// 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 have multiple inputs with different IDs, but
+	// the port number must match our ControlPort()
+	if (dest.port != ControlPort())
+		return B_MEDIA_BAD_DESTINATION;
+
+	// specialize to raw audio format if necessary
+	if (ioFormat->type == B_MEDIA_UNKNOWN_TYPE)
+		ioFormat->type = B_MEDIA_RAW_AUDIO;
+
+	// we require a raw audio format
+	if (ioFormat->type != B_MEDIA_RAW_AUDIO)
+		return B_MEDIA_BAD_FORMAT;
+
+	// We do not have special requirements, but just in case
+	// another mixer is connecting to us and may need a hint
+	// to create a connection at optimal frame rate and
+	// channel count, we place this information in the user_data
+	fCore->Lock();
+	MixerOutput *output = fCore->Output();
+	uint32 channel_count = output ? output->MediaOutput().format.u.raw_audio.channel_count : 0;
+	float frame_rate = output ? output->MediaOutput().format.u.raw_audio.frame_rate : 0.0;
+	fCore->Unlock();
+	ioFormat->user_data_type = FORMAT_USER_DATA_TYPE;
+	*(uint32 *)&ioFormat->user_data[0] = FORMAT_USER_DATA_MAGIC_1;
+	*(uint32 *)&ioFormat->user_data[4] = channel_count;
+	*(float *)&ioFormat->user_data[20] = frame_rate;
+	*(uint32 *)&ioFormat->user_data[44] = FORMAT_USER_DATA_MAGIC_2;
+
+	return B_OK;
+}
+
+
+status_t
+AudioMixer::GetNextInput(int32 *cookie, media_input *out_input)
+{
+	TRACE("AudioMixer::GetNextInput\n");
+
+	// our 0th input is always a wildcard and free one
+	if (*cookie == 0) {
+		out_input->node = Node();
+		out_input->source = media_source::null;
+		out_input->destination.port = ControlPort();
+		out_input->destination.id = 0;
+		memset(&out_input->format, 0, sizeof(out_input->format));
+		out_input->format.type = B_MEDIA_RAW_AUDIO;
+		strcpy(out_input->name, "Free Input");
+		*cookie += 1;
+		return B_OK;
+	}
+
+	// the other inputs are the currently connected ones
+	fCore->Lock();
+	MixerInput *input;
+	input = fCore->Input(*cookie - 1);
+	if (!input) {
+		fCore->Unlock();
+		return B_BAD_INDEX;
+	}
+	*out_input = input->MediaInput();
+	*cookie += 1;
+	fCore->Unlock();
+	return B_OK;
+}
+
+
+void
+AudioMixer::DisposeInputCookie(int32 cookie)
+{
+	// nothing to do
+}
+
+
+void
+AudioMixer::BufferReceived(BBuffer *buffer)
+{
+
+	if (buffer->Header()->type == B_MEDIA_PARAMETERS) {
+		TRACE("Control Buffer Received\n");
+		ApplyParameterData(buffer->Data(), buffer->SizeUsed());
+		buffer->Recycle();
+		return;
+	}
+
+	//PRINT(4, "buffer received at %12Ld, should arrive at %12Ld, delta %12Ld\n", TimeSource()->Now(), buffer->Header()->start_time, TimeSource()->Now() - buffer->Header()->start_time);
+
+	// Note: The following code is outcommented on purpose
+	// and is about to be modified at a later point
+	//	HandleInputBuffer(buffer, 0);
+	//	buffer->Recycle();
+	//	return;
+
+
+	// 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, bigtime_t lateness)
+{
+	// Note: The following code is outcommented on purpose
+	// and is about to be modified at a later point
+	/*
+	if (lateness > 5000) {
+		printf("Received buffer with way to high lateness %Ld\n", lateness);
+		if (RunMode() != B_DROP_DATA) {
+			printf("sending notify\n");
+			NotifyLateProducer(channel->fInput.source, lateness / 2, TimeSource()->Now());
+		} else if (RunMode() == B_DROP_DATA) {
+			printf("dropping buffer\n");
+			return;
+		}
+	}
+	*/
+
+	//	printf("Received buffer with lateness %Ld\n", lateness);
+
+	fCore->Lock();
+	fCore->BufferReceived(buffer, lateness);
+	fCore->Unlock();
+
+	// Note: The following code is outcommented on purpose
+	// and is about to be modified at a later point
+	/*
+		if ((B_OFFLINE == RunMode()) && (B_DATA_AVAILABLE == channel->fProducerDataStatus))
+		{
+			RequestAdditionalBuffer(channel->fInput.source, buffer);
+		}
+	*/
+}
+
+
+void
+AudioMixer::ProducerDataStatus(const media_destination& for_whom,
+	int32 status, bigtime_t at_performance_time)
+{
+/*
+	if (IsValidDest(for_whom))
+	{
+		media_timed_event event(at_performance_time, BTimedEventQueue::B_DATA_STATUS,
+			(void *)(&for_whom), BTimedEventQueue::B_NO_CLEANUP, status, 0, NULL);
+		EventQueue()->AddEvent(event);
+
+		// FIX_THIS
+		// the for_whom destination is not being sent correctly - verify in HandleEvent loop
+
+	}
+*/
+}
+
+
+status_t
+AudioMixer::GetLatencyFor(const media_destination &for_whom,
+	bigtime_t *out_latency, media_node_id *out_timesource)
+{
+	// we have multiple inputs with different IDs, but
+	// the port number must match our ControlPort()
+	if (for_whom.port != ControlPort())
+		return B_MEDIA_BAD_DESTINATION;
+
+	// return our event latency - this includes our internal + downstream
+	// latency, but _not_ the scheduling latency
+	*out_latency = EventLatency();
+	*out_timesource = TimeSource()->ID();
+
+	printf("AudioMixer::GetLatencyFor %Ld, timesource is %ld\n",
+		*out_latency, *out_timesource);
+
+	return B_OK;
+}
+
+
+status_t
+AudioMixer::Connected(const media_source &producer,
+	const media_destination &where, const media_format &with_format,
+	media_input *out_input)
+{
+	PRINT_FORMAT("AudioMixer::Connected: ", with_format);
+
+	// workaround for a crashing bug in RealPlayer.  to be proper, RealPlayer's
+	// BBufferProducer::PrepareToConnect() should have removed all wildcards.
+	if (out_input->format.u.raw_audio.frame_rate == 0) {
+		fprintf(stderr, "Audio Mixer Warning: "
+			"Producer (port %ld, id %ld) connected with frame_rate=0\n",
+			producer.port, producer.id);
+		MixerOutput *output = fCore->Output();
+		float frame_rate = output
+			? output->MediaOutput().format.u.raw_audio.frame_rate : 44100.0f;
+		out_input->format.u.raw_audio.frame_rate = frame_rate;
+	}
+
+	// a BBufferProducer is connecting to our BBufferConsumer
+
+	// incoming connections should always have an incoming ID=0,
+	// and the port number must match our ControlPort()
+	if (where.id != 0 || where.port != ControlPort())
+		return B_MEDIA_BAD_DESTINATION;
+
+	fCore->Lock();
+
+	// we assign a new id (!= 0) to the newly created input
+	out_input->destination.id = fCore->CreateInputID();
+
+	// 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 the input has no name, assign one
+	if (strlen(out_input->name) == 0)
+		sprintf(out_input->name, "Input %ld", out_input->destination.id);
+
+	// add a new input to the mixer engine
+	MixerInput *input;
+	input = fCore->AddInput(*out_input);
+
+	fCore->Settings()->LoadConnectionSettings(input);
+
+	fCore->Unlock();
+
+	// 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.
+	// But we have no special buffer requirements anyway...
+
+	UpdateParameterWeb();
+
+	return B_OK;
+}
+
+
+void
+AudioMixer::Disconnected(const media_source &producer,
+	const media_destination &where)
+{
+	// One of our inputs has been disconnected
+
+	// check if it is really belongs to us
+	if (where.port != ControlPort()) {
+		TRACE("AudioMixer::Disconnected wrong input port\n");
+		return;
+	}
+
+	fCore->Lock();
+
+	if (!fCore->RemoveInput(where.id)) {
+		TRACE("AudioMixer::Disconnected can't remove input\n");
+	}
+
+	fCore->Unlock();
+	UpdateParameterWeb();
+}
+
+
+status_t
+AudioMixer::FormatChanged(const media_source &producer,
+	const media_destination &consumer, int32 change_tag,
+	const media_format &format)
+{
+	// at some point in the future (indicated by change_tag and RequestCompleted()),
+	// we will receive buffers in a different format
+
+	TRACE("AudioMixer::FormatChanged\n");
+
+	if (consumer.port != ControlPort() || consumer.id == 0)
+		return B_MEDIA_BAD_DESTINATION;
+
+	if (fCore->Settings()->RefuseInputFormatChange()) {
+		TRACE("AudioMixer::FormatChanged: input format change refused\n");
+		return B_ERROR;
+	}
+
+	// XXX we should not apply the format change at this point
+
+	// tell core about format change
+	fCore->Lock();
+	fCore->InputFormatChanged(consumer.id, format.u.raw_audio);
+	fCore->Unlock();
+
+	return B_OK;
+}
+
+
+//	#pragma mark - BBufferProducer methods
+
+
+status_t
+AudioMixer::FormatSuggestionRequested(media_type type, int32 quality,
+	media_format *format)
+{
+	TRACE("AudioMixer::FormatSuggestionRequested\n");
+
+	// BBufferProducer function, a downstream consumer
+	// is asking for our output format
+
+	if (type != B_MEDIA_RAW_AUDIO && type != B_MEDIA_UNKNOWN_TYPE)
+		return B_MEDIA_BAD_FORMAT;
+
+	// we can produce any (wildcard) raw audio format
+	memset(format, 0, sizeof(*format));
+	format->type = B_MEDIA_RAW_AUDIO;
+	return B_OK;
+}
+
+
+status_t
+AudioMixer::FormatProposal(const media_source &output, media_format *ioFormat)
+{
+	// BBufferProducer function, we implement this function to verify that the
+	// proposed media_format is suitable for the specified output. If any fields
+	// in the format are wildcards, and we have a specific requirement, adjust
+	// those fields to match our requirements before returning.
+
+	TRACE("AudioMixer::FormatProposal\n");
+
+	// we only have one output (id=0, port=ControlPort())
+	if (output.id != 0 || output.port != ControlPort())
+		return B_MEDIA_BAD_SOURCE;
+
+	// specialize to raw audio format if necessary
+	if (ioFormat->type == B_MEDIA_UNKNOWN_TYPE)
+		ioFormat->type = B_MEDIA_RAW_AUDIO;
+
+	// we require a raw audio format
+	if (ioFormat->type != B_MEDIA_RAW_AUDIO)
+		return B_MEDIA_BAD_FORMAT;
+
+	return B_OK;
+}
+
+/*!	If the format isn't good, put a good format into *io_format and return error
+	If format has wildcard, specialize to what you can do (and change).
+	If you can change the format, return OK.
+	The request comes from your destination sychronously, so you cannot ask it
+	whether it likes it -- you should assume it will since it asked.
+*/
+status_t
+AudioMixer::FormatChangeRequested(const media_source &source,
+	const media_destination &destination, media_format *io_format,
+	int32 *_deprecated_)
+{
+	// the downstream consumer node (soundcard) requested that we produce
+	// another format, we need to check if the format is acceptable and
+	// remove any wildcards before returning OK.
+
+	TRACE("AudioMixer::FormatChangeRequested\n");
+
+	if (fCore->Settings()->RefuseOutputFormatChange()) {
+		TRACE("AudioMixer::FormatChangeRequested: output format change refused\n");
+		return B_ERROR;
+	}
+
+	fCore->Lock();
+
+	MixerOutput *output = fCore->Output();
+	if (!output) {
+		ERROR("AudioMixer::FormatChangeRequested: no output\n");
+		goto err;
+	}
+	if (source != output->MediaOutput().source) {
+		ERROR("AudioMixer::FormatChangeRequested: wrong output source\n");
+		goto err;
+	}
+	if (destination != output->MediaOutput().destination) {
+		ERROR("AudioMixer::FormatChangeRequested: wrong output destination "
+			"(port %ld, id %ld), our is (port %ld, id %ld)\n", destination.port,
+			destination.id, output->MediaOutput().destination.port,
+			output->MediaOutput().destination.id);
+		if (destination.port == output->MediaOutput().destination.port
+			&& destination.id == output->MediaOutput().destination.id + 1) {
+			ERROR("AudioMixer::FormatChangeRequested: this might be the broken "
+				"R5 multi audio add-on\n");
+		}
+		goto err;
+	}
+	if (io_format->type != B_MEDIA_RAW_AUDIO
+		&& io_format->type != B_MEDIA_UNKNOWN_TYPE) {
+		ERROR("AudioMixer::FormatChangeRequested: wrong format type\n");
+		goto err;
+	}
+
+	/* remove wildcards */
+#if USE_MEDIA_FORMAT_WORKAROUND
+	multi_audio_format_specialize(&io_format->u.raw_audio,
+		&fDefaultFormat.u.raw_audio);
+#else
+	io_format->SpecializeTo(&fDefaultFormat);
+#endif
+
+	media_node_id id;
+	FindLatencyFor(destination, &fDownstreamLatency, &id);
+	TRACE("AudioMixer: Downstream Latency is %Ld usecs\n", fDownstreamLatency);
+
+	// SetDuration of one buffer
+	SetBufferDuration(buffer_duration(io_format->u.raw_audio));
+	TRACE("AudioMixer: buffer duration is %Ld usecs\n", BufferDuration());
+
+	// Our internal latency is at least the length of a full output buffer
+	fInternalLatency = BufferDuration()
+		+ max(4500LL, bigtime_t(0.5 * BufferDuration()));
+	TRACE("AudioMixer: Internal latency is %Ld usecs\n", fInternalLatency);
+
+	SetEventLatency(fDownstreamLatency + fInternalLatency);
+
+	// we need to inform all connected *inputs* about *our* change in latency
+	PublishEventLatencyChange();
+
+	// TODO: we might need to create more buffers, to span a larger downstream
+	// latency
+
+	// apply latency change
+	fCore->SetTimingInfo(TimeSource(), fDownstreamLatency);
+
+	// apply format change
+	fCore->OutputFormatChanged(io_format->u.raw_audio);
+
+	delete fBufferGroup;
+	fBufferGroup = CreateBufferGroup();
+	fCore->SetOutputBufferGroup(fBufferGroup);
+
+	fCore->Unlock();
+	return B_OK;
+
+err:
+	fCore->Unlock();
+	return B_ERROR;
+}
+
+
+status_t
+AudioMixer::GetNextOutput(int32 *cookie, media_output *out_output)
+{
+	TRACE("AudioMixer::GetNextOutput\n");
+
+	if (*cookie != 0)
+		return B_BAD_INDEX;
+
+	fCore->Lock();
+	MixerOutput *output = fCore->Output();
+	if (output) {
+		*out_output = output->MediaOutput();
+	} else {
+		out_output->node = Node();
+		out_output->source.port = ControlPort();
+		out_output->source.id = 0;
+		out_output->destination = media_destination::null;
+		memset(&out_output->format, 0, sizeof(out_output->format));
+		out_output->format.type = B_MEDIA_RAW_AUDIO;
+		strcpy(out_output->name, "Mixer Output");
+	}
+	fCore->Unlock();
+
+	*cookie += 1;
+	return B_OK;
+}
+
+
+status_t
+AudioMixer::DisposeOutputCookie(int32 cookie)
+{
+	// nothing to do
+	return B_OK;
+}
+
+
+status_t
+AudioMixer::SetBufferGroup(const media_source &for_source,
+	BBufferGroup *newGroup)
+{
+	PRINT("AudioMixer::SetBufferGroup\n");
+	// the downstream consumer (soundcard) node asks us to use another
+	// BBufferGroup (might be NULL). We only have one output (id 0)
+	if (for_source.port != ControlPort() || for_source.id != 0)
+		return B_MEDIA_BAD_SOURCE;
+
+	if (newGroup == fBufferGroup) {
+		// we're already using this buffergroup
+		return B_OK;
+	}
+
+	fCore->Lock();
+	if (!newGroup)
+		newGroup = CreateBufferGroup();
+	fCore->SetOutputBufferGroup(newGroup);
+	delete fBufferGroup;
+	fBufferGroup = newGroup;
+	fCore->Unlock();
+
+	return B_OK;
+}
+
+
+status_t
+AudioMixer::GetLatency(bigtime_t *out_latency)
+{
+	// report our *total* latency:  internal plus downstream plus scheduling
+	*out_latency = EventLatency() + SchedulingLatency();
+
+	TRACE("AudioMixer::GetLatency %Ld\n", *out_latency);
+
+	return B_OK;
+}
+
+
+void
+AudioMixer::LatencyChanged(const media_source& source,
+	const media_destination& destination, bigtime_t new_latency, uint32 flags)
+{
+	if (source.port != ControlPort() || source.id != 0) {
+		ERROR("AudioMixer::LatencyChanged: received but has wrong source "
+			"%ld/%ld\n", source.port, source.id);
+		return;
+	}
+
+	TRACE("AudioMixer::LatencyChanged: downstream latency from %ld/%ld to "
+		"%ld/%ld changed from %Ld to %Ld\n", source.port, source.id,
+		destination.port, destination.id, fDownstreamLatency, new_latency);
+
+#if DEBUG
+	{
+		media_node_id id;
+		bigtime_t l;
+		FindLatencyFor(destination, &l, &id);
+		TRACE("AudioMixer: Reported downstream Latency is %Ld usecs\n", l);
+	}
+#endif
+
+	fDownstreamLatency = new_latency;
+	SetEventLatency(fDownstreamLatency + fInternalLatency);
+
+	// XXX we might need to create more buffers, to span a larger downstream
+	// latency
+
+	fCore->Lock();
+	fCore->SetTimingInfo(TimeSource(), fDownstreamLatency);
+	PublishEventLatencyChange();
+	fCore->Unlock();
+}
+
+status_t
+AudioMixer::PrepareToConnect(const media_source &what,
+	const media_destination &where, media_format *format,
+	media_source *out_source, char *out_name)
+{
+	TRACE("AudioMixer::PrepareToConnect\n");
+	// PrepareToConnect() is the second stage of format negotiations that
+	// happens inside BMediaRoster::Connect(). At this point, the consumer's
+	// AcceptFormat() method has been called, and that node has potentially
+	// changed the proposed format.
+	// It may also have left wildcards in the format. PrepareToConnect()
+	// *must* fully specialize the format before returning!
+	// we also create the new output connection and return it in out_source.
+
+	PRINT_FORMAT("AudioMixer::PrepareToConnect: suggested format", *format);
+
+	// avoid loop connections
+	if (where.port == ControlPort())
+		return B_MEDIA_BAD_SOURCE;
+
+	// is the source valid?
+	if (what.port != ControlPort() || what.id != 0)
+		return B_MEDIA_BAD_SOURCE;
+
+	// is the format acceptable?
+	if (format->type != B_MEDIA_RAW_AUDIO
+		&& format->type != B_MEDIA_UNKNOWN_TYPE) {
+		PRINT_FORMAT("AudioMixer::PrepareToConnect: bad format", *format);
+		return B_MEDIA_BAD_FORMAT;
+	}
+
+	fCore->Lock();
+
+	// are we already connected?
+	if (fCore->Output() != 0) {
+		fCore->Unlock();
+		ERROR("AudioMixer::PrepareToConnect: already connected\n");
+		return B_MEDIA_ALREADY_CONNECTED;
+	}
+
+	// It is possible that another mixer is connecting.
+	// To avoid using the default format, we use one of
+	// a) the format that it indicated as hint in the user_data,
+	// b) the output format of the system audio mixer
+	// c) the input format of the system DAC device
+	// d) if everything failes, keep the wildcard
+	if (format->u.raw_audio.channel_count == 0
+		&& format->u.raw_audio.frame_rate < 1
+		&& format->user_data_type == FORMAT_USER_DATA_TYPE
+		&& *(uint32 *)&format->user_data[0] == FORMAT_USER_DATA_MAGIC_1
+		&& *(uint32 *)&format->user_data[44] == FORMAT_USER_DATA_MAGIC_2) {
+		// ok, a mixer is connecting
+		uint32 channel_count = *(uint32 *)&format->user_data[4];
+		float frame_rate = *(float *)&format->user_data[20];
+		if (channel_count > 0 && frame_rate > 0) {
+			// format is good, use it
+			format->u.raw_audio.channel_count = channel_count;
+			format->u.raw_audio.frame_rate = frame_rate;
+		} else {
+			// other mixer's output is probably not connected
+			media_node node;
+			BMediaRoster *roster = BMediaRoster::Roster();
+			media_output out;
+			media_input in;
+			int32 count;
+			if (roster->GetAudioMixer(&node) == B_OK
+				&& roster->GetConnectedOutputsFor(node, &out, 1, &count)
+						== B_OK
+				&& count == 1) {
+				// use mixer output format
+				format->u.raw_audio.channel_count
+					= out.format.u.raw_audio.channel_count;
+				format->u.raw_audio.frame_rate
+					= out.format.u.raw_audio.frame_rate;
+			} else if (roster->GetAudioOutput(&node) == B_OK
+				&& roster->GetAllInputsFor(node, &in, 1, &count) == B_OK
+				&& count == 1) {
+				// use DAC input format
+				format->u.raw_audio.channel_count
+					= in.format.u.raw_audio.channel_count;
+				format->u.raw_audio.frame_rate
+					= in.format.u.raw_audio.frame_rate;
+			}
+		}
+	}
+
+	/* set source and suggest a name */
+	*out_source = what;
+	strcpy(out_name, "Mixer Output");
+
+	/* remove wildcards */
+#if USE_MEDIA_FORMAT_WORKAROUND
+	multi_audio_format_specialize(&format->u.raw_audio,
+		&fDefaultFormat.u.raw_audio);
+#else
+	format->SpecializeTo(&fDefaultFormat);
+#endif
+
+	PRINT_FORMAT("AudioMixer::PrepareToConnect: final format", *format);
+
+	/* add output to core */
+	media_output output;
+	output.node = Node();
+	output.source = *out_source;
+	output.destination = where;
+	output.format = *format;
+	strcpy(output.name, out_name);
+
+	fCore->EnableOutput(false);
+	fCore->AddOutput(output);
+
+	fCore->Unlock();
+	return B_OK;
+}
+
+
+void
+AudioMixer::Connect(status_t error, const media_source &source,
+	const media_destination &dest, const media_format &format, char *io_name)
+{
+	TRACE("AudioMixer::Connect\n");
+
+	fCore->Lock();
+	// are we still connected?
+	if (fCore->Output() == 0) {
+		fCore->Unlock();
+		ERROR("AudioMixer::Connect: no longer connected\n");
+		return;
+	}
+	fCore->Unlock();
+
+	if (error != B_OK) {
+		// if an error occured, remove output from core
+		ERROR("AudioMixer::Connect failed with error 0x%08lX, removing "
+			"connection\n", error);
+		fCore->Lock();
+		fCore->RemoveOutput();
+		fCore->Unlock();
+		return;
+	}
+
+	/* Switch our prefered format to have the same
+	 * frame_rate and channel count as the output.
+	 */
+	fDefaultFormat.u.raw_audio.frame_rate = format.u.raw_audio.frame_rate;
+	fDefaultFormat.u.raw_audio.channel_count = format.u.raw_audio.channel_count;
+
+	// if the connection has no name, we set it now
+	if (strlen(io_name) == 0)
+		strcpy(io_name, "Mixer Output");
+
+	// Now that we're connected, we can determine our downstream latency.
+	media_node_id id;
+	FindLatencyFor(dest, &fDownstreamLatency, &id);
+	TRACE("AudioMixer: Downstream Latency is %Ld usecs\n", fDownstreamLatency);
+
+	// SetDuration of one buffer
+	SetBufferDuration(buffer_duration(format.u.raw_audio));
+	TRACE("AudioMixer: buffer duration is %Ld usecs\n", BufferDuration());
+
+	// Our internal latency is at least the length of a full output buffer
+	fInternalLatency = BufferDuration()
+		+ max(4500LL, bigtime_t(0.5 * BufferDuration()));
+	TRACE("AudioMixer: Internal latency is %Ld usecs\n", fInternalLatency);
+
+	SetEventLatency(fDownstreamLatency + fInternalLatency);
+
+	// we need to inform all connected *inputs* about *our* change in latency
+	PublishEventLatencyChange();
+
+	// 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)
+		fBufferGroup = CreateBufferGroup();
+
+	fCore->Lock();
+
+	ASSERT(fCore->Output() != 0);
+
+	// our source should still be valid, too
+	ASSERT(fCore->Output()->MediaOutput().source.id == 0);
+	ASSERT(fCore->Output()->MediaOutput().source.port == ControlPort());
+
+	// BBufferConsumer::Connected() may return a different input for the
+	// newly created connection. The destination can have changed since
+	// AudioMixer::PrepareToConnect() and we need to update it.
+	fCore->Output()->MediaOutput().destination = dest;
+
+	fCore->EnableOutput(true);
+	fCore->SetTimingInfo(TimeSource(), fDownstreamLatency);
+	fCore->SetOutputBufferGroup(fBufferGroup);
+
+	fCore->Settings()->LoadConnectionSettings(fCore->Output());
+
+	fCore->Unlock();
+	UpdateParameterWeb();
+}
+
+void
+AudioMixer::Disconnect(const media_source &what, const media_destination &where)
+{
+	TRACE("AudioMixer::Disconnect\n");
+	fCore->Lock();
+
+	// Make sure that our connection is the one being disconnected
+	MixerOutput * output = fCore->Output();
+	if (!output
+		|| output->MediaOutput().node != Node()
+		|| output->MediaOutput().source != what
+		|| output->MediaOutput().destination != where) {
+		ERROR("AudioMixer::Disconnect can't disconnect (wrong connection)\n");
+		fCore->Unlock();
+		return;
+	}
+
+	// Switch our prefered format back to default
+	// frame rate and channel count.
+	fDefaultFormat.u.raw_audio.frame_rate = 96000;
+	fDefaultFormat.u.raw_audio.channel_count = 2;
+
+	// force a stop
+	fCore->Stop();
+
+	fCore->RemoveOutput();
+
+	// destroy buffer group
+	delete fBufferGroup;
+	fBufferGroup = 0;
+	fCore->SetOutputBufferGroup(0);
+
+	fCore->Unlock();
+	UpdateParameterWeb();
+}
+
+
+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
+
+	ERROR("AudioMixer::LateNoticeReceived, %Ld too late at %Ld\n", how_much,
+		performance_time);
+	// Note: The following code is outcommented on purpose
+	// and is about to be modified at a later point
+	/*
+	if (what == fOutput.source) {
+		if (RunMode() == B_INCREASE_LATENCY) {
+			fInternalLatency += how_much;
+
+			if (fInternalLatency > 50000)
+				fInternalLatency = 50000;
+
+			printf("AudioMixer: increasing internal latency to %Ld usec\n", fInternalLatency);
+			SetEventLatency(fDownstreamLatency + fInternalLatency);
+
+			PublishEventLatencyChange();
+		}
+	}
+	*/
+}
+
+
+void
+AudioMixer::EnableOutput(const media_source &what, bool enabled,
+	int32 *_deprecated_)
+{
+	// we only have one output
+	if (what.id != 0 || what.port != ControlPort())
+		return;
+
+	fCore->Lock();
+	fCore->EnableOutput(enabled);
+	fCore->Unlock();
+}
+
+
+//	#pragma mark - BMediaEventLooper methods
+
+
+void
+AudioMixer::NodeRegistered()
+{
+	Run();
+	SetPriority(120);
+	UpdateParameterWeb();
+}
+
+
+void
+AudioMixer::SetTimeSource(BTimeSource * time_source)
+{
+	TRACE("AudioMixer::SetTimeSource: timesource is now %ld\n",
+		time_source->ID());
+	fCore->Lock();
+	fCore->SetTimingInfo(time_source, fDownstreamLatency);
+	fCore->Unlock();
+}
+
+
+void
+AudioMixer::HandleEvent(const media_timed_event *event, bigtime_t lateness,
+	bool realTimeEvent)
+{
+	switch (event->type) {
+		case BTimedEventQueue::B_HANDLE_BUFFER:
+		{
+			HandleInputBuffer((BBuffer *)event->pointer, lateness);
+			((BBuffer *)event->pointer)->Recycle();
+			break;
+		}
+
+		case BTimedEventQueue::B_START:
+		{
+			TRACE("AudioMixer::HandleEvent: B_START\n");
+			if (RunState() != B_STARTED) {
+				fCore->Lock();
+				fCore->Start();
+				fCore->Unlock();
+			}
+			break;
+		}
+
+		case BTimedEventQueue::B_STOP:
+		{
+			TRACE("AudioMixer::HandleEvent: B_STOP\n");
+			// stopped - don't process any more buffers, flush all buffers
+			// from event queue
+			EventQueue()->FlushEvents(0, BTimedEventQueue::B_ALWAYS, true,
+				BTimedEventQueue::B_HANDLE_BUFFER);
+			fCore->Lock();
+			fCore->Stop();
+			fCore->Unlock();
+			break;
+		}
+
+		case BTimedEventQueue::B_DATA_STATUS:
+		{
+			ERROR("DataStatus message\n");
+			break;
+		}
+
+		default:
+			break;
+	}
+}
+
+
+//	#pragma mark - AudioMixer methods
+
+
+void
+AudioMixer::PublishEventLatencyChange()
+{
+	// our event (processing + downstream) latency has changed,
+	// and we need tell all inputs about this
+
+	TRACE("AudioMixer::PublishEventLatencyChange\n");
+
+	fCore->Lock();
+
+	MixerInput *input;
+	for (int i = 0; (input = fCore->Input(i)) != 0; i++) {
+		TRACE("AudioMixer::PublishEventLatencyChange: SendLatencyChange, "
+			"connection %ld/%ld to %ld/%ld event latency is now %Ld\n",
+			input->MediaInput().source.port, input->MediaInput().source.id,
+			input->MediaInput().destination.port,
+			input->MediaInput().destination.id, EventLatency());
+		SendLatencyChange(input->MediaInput().source,
+			input->MediaInput().destination, EventLatency());
+	}
+
+	fCore->Unlock();
+}
+
+
+BBufferGroup*
+AudioMixer::CreateBufferGroup()
+{
+	// allocate enough buffers to span our downstream latency
+	// (plus one for rounding up), plus one extra
+	int32 count = int32(fDownstreamLatency / BufferDuration()) + 2;
+
+	TRACE("AudioMixer::CreateBufferGroup: fDownstreamLatency %Ld, "
+		"BufferDuration %Ld, buffer count = %ld\n", fDownstreamLatency,
+		BufferDuration(), count);
+
+	if (count < 3)
+		count = 3;
+
+	fCore->Lock();
+	uint32 size = fCore->Output()->MediaOutput().format.u.raw_audio.buffer_size;
+	fCore->Unlock();
+
+	TRACE("AudioMixer: allocating %ld buffers of %ld bytes each\n",
+		count, size);
+	return new BBufferGroup(size, count);
+}
+
+
+float
+AudioMixer::dB_to_Gain(float db)
+{
+	TRACE("dB_to_Gain: dB in: %01.2f ", db);
+	if (fCore->Settings()->NonLinearGainSlider()) {
+		if (db > 0) {
+			db = db * (pow(abs(DB_MAX), (1.0 / DB_EXPONENT_POSITIVE))
+				/ abs(DB_MAX));
+			db = pow(db, DB_EXPONENT_POSITIVE);
+		} else {
+			db = -db;
+			db = db * (pow(abs(DB_MIN), (1.0 / DB_EXPONENT_NEGATIVE))
+				/ abs(DB_MIN));
+			db = pow(db, DB_EXPONENT_NEGATIVE);
+			db = -db;
+		}
+	}
+	TRACE("dB out: %01.2f\n", db);
+	return pow(10.0, db / 20.0);
+}
+
+
+float
+AudioMixer::Gain_to_dB(float gain)
+{
+	float db;
+	db = 20.0 * log10(gain);
+	if (fCore->Settings()->NonLinearGainSlider()) {
+		if (db > 0) {
+			db = pow(db, (1.0 / DB_EXPONENT_POSITIVE));
+			db = db * (abs(DB_MAX) / pow(abs(DB_MAX),
+				(1.0 / DB_EXPONENT_POSITIVE)));
+		} else {
+			db = -db;
+			db = pow(db, (1.0 / DB_EXPONENT_NEGATIVE));
+			db = db * (abs(DB_MIN) / pow(abs(DB_MIN),
+				(1.0 / DB_EXPONENT_NEGATIVE)));
+			db = -db;
+		}
+	}
+	return db;
+}
+
+
+//	#pragma markß - BControllable methods
+
+
+status_t
 AudioMixer::GetParameterValue(int32 id, bigtime_t *last_change, void *value,
 	size_t *ioSize)
 {
@@ -1570,7 +1639,7 @@ AudioMixer::SetParameterValue(int32 id, bigtime_t when, const void *value,
 			} else {
 				input->RemoveInputChannelDestination(PARAM_CHAN(id), PARAM_DST(id));
 			}
-			// XXX this is really annoying
+			// TODO: this is really annoying
 			// The slider count of the gain control needs to be changed,
 			// but calling SetChannelCount(input->GetMixerChannelCount())
 			// on it has no effect on remote BParameterWebs in other apps.
@@ -1582,18 +1651,21 @@ AudioMixer::SetParameterValue(int32 id, bigtime_t when, const void *value,
 		}
 		fCore->Settings()->SaveConnectionSettings(input);
 	}
+
 	BroadcastNewParameterValue(when, id, const_cast<void *>(value), size);
+
 err:
 	fCore->Unlock();
 	if (update)
 		UpdateParameterWeb();
 }
 
+
 void
 AudioMixer::UpdateParameterWeb()
 {
 	fCore->Lock();
-	BParameterWeb *web = new BParameterWeb(); 
+	BParameterWeb *web = new BParameterWeb();
 	BParameterGroup *top;
 	BParameterGroup *outputchannels;
 	BParameterGroup *inputchannels;
@@ -1604,12 +1676,12 @@ AudioMixer::UpdateParameterWeb()
 	MixerInput *in;
 	MixerOutput *out;
 	char buf[50];
-	
+
 	top = web->MakeGroup("Gain Controls");
 
 	out = fCore->Output();
 	group = top->MakeGroup("");
-	group->MakeNullParameter(PARAM_STR1(0), B_MEDIA_RAW_AUDIO, "Master Output", B_WEB_BUFFER_INPUT); 
+	group->MakeNullParameter(PARAM_STR1(0), B_MEDIA_RAW_AUDIO, "Master Output", B_WEB_BUFFER_INPUT);
 	if (!out) {
 		group->MakeNullParameter(PARAM_STR2(0), B_MEDIA_RAW_AUDIO, "not connected", B_GENERIC);
 	} else {
@@ -1622,14 +1694,14 @@ AudioMixer::UpdateParameterWeb()
 		} else {
 			// multi channel control
 			group->MakeContinuousParameter(PARAM_GAIN(0), B_MEDIA_RAW_AUDIO, "Gain", B_MASTER_GAIN, "dB", DB_MIN, DB_MAX, 0.1)
-										   ->SetChannelCount(out->GetOutputChannelCount()); 
+										   ->SetChannelCount(out->GetOutputChannelCount());
 		}
-		group->MakeNullParameter(PARAM_STR3(0), B_MEDIA_RAW_AUDIO, "To Output", B_WEB_BUFFER_OUTPUT); 
+		group->MakeNullParameter(PARAM_STR3(0), B_MEDIA_RAW_AUDIO, "To Output", B_WEB_BUFFER_OUTPUT);
 	}
 
 	for (int i = 0; (in = fCore->Input(i)); i++) {
 		group = top->MakeGroup("");
-		group->MakeNullParameter(PARAM_STR1(in->ID()), B_MEDIA_RAW_AUDIO, in->MediaInput().name, B_WEB_BUFFER_INPUT); 
+		group->MakeNullParameter(PARAM_STR1(in->ID()), B_MEDIA_RAW_AUDIO, in->MediaInput().name, B_WEB_BUFFER_INPUT);
 		group->MakeNullParameter(PARAM_STR2(in->ID()), B_MEDIA_RAW_AUDIO, StringForFormat(buf, in), B_GENERIC);
 		group->MakeDiscreteParameter(PARAM_MUTE(in->ID()), B_MEDIA_RAW_AUDIO, "Mute", B_MUTE);
 		// XXX the gain control is ugly once you have more than two channels,
@@ -1642,7 +1714,7 @@ AudioMixer::UpdateParameterWeb()
 				group->MakeContinuousParameter(PARAM_BALANCE(in->ID()), B_MEDIA_RAW_AUDIO, "", B_BALANCE, "", 0, 100, 1);
 			} else {
 				// multi channel control
-				group->MakeContinuousParameter(PARAM_GAIN(in->ID()), B_MEDIA_RAW_AUDIO, "Gain", B_GAIN, "dB", DB_MIN, DB_MAX, 0.1) 
+				group->MakeContinuousParameter(PARAM_GAIN(in->ID()), B_MEDIA_RAW_AUDIO, "Gain", B_GAIN, "dB", DB_MIN, DB_MAX, 0.1)
 											   ->SetChannelCount(in->GetInputChannelCount());
 			}
 		} else {
@@ -1653,38 +1725,43 @@ AudioMixer::UpdateParameterWeb()
 				group->MakeContinuousParameter(PARAM_BALANCE(in->ID()), B_MEDIA_RAW_AUDIO, "", B_BALANCE, "", 0, 100, 1);
 			} else {
 				// multi channel control
-				group->MakeContinuousParameter(PARAM_GAIN(in->ID()), B_MEDIA_RAW_AUDIO, "Gain", B_GAIN, "dB", DB_MIN, DB_MAX, 0.1) 
+				group->MakeContinuousParameter(PARAM_GAIN(in->ID()), B_MEDIA_RAW_AUDIO, "Gain", B_GAIN, "dB", DB_MIN, DB_MAX, 0.1)
 											   ->SetChannelCount(in->GetMixerChannelCount());
 			}
 		}
-		group->MakeNullParameter(PARAM_STR3(in->ID()), B_MEDIA_RAW_AUDIO, "To Master", B_WEB_BUFFER_OUTPUT); 
+		group->MakeNullParameter(PARAM_STR3(in->ID()), B_MEDIA_RAW_AUDIO, "To Master", B_WEB_BUFFER_OUTPUT);
 	}
-	
+
 	if (fCore->Settings()->AllowOutputChannelRemapping()) {
 		top = web->MakeGroup("Output Mapping"); // top level group
 		outputchannels = top->MakeGroup("");
 		outputchannels->MakeNullParameter(PARAM_STR4(0), B_MEDIA_RAW_AUDIO, "Output Channel Sources", B_GENERIC);
 
 		group = outputchannels->MakeGroup("");
-		group->MakeNullParameter(PARAM_STR5(0), B_MEDIA_RAW_AUDIO, "Master Output", B_GENERIC); 
+		group->MakeNullParameter(PARAM_STR5(0), B_MEDIA_RAW_AUDIO, "Master Output", B_GENERIC);
 		group = group->MakeGroup("");
 		if (!out) {
 			group->MakeNullParameter(PARAM_STR6(0), B_MEDIA_RAW_AUDIO, "not connected", B_GENERIC);
 		} else {
 			for (int chan = 0; chan < out->GetOutputChannelCount(); chan++) {
 				subgroup = group->MakeGroup("");
-				subgroup->MakeNullParameter(PARAM_SRC_STR(0, chan), B_MEDIA_RAW_AUDIO,
-											StringForChannelType(buf, out->GetOutputChannelType(chan)), B_GENERIC);
+				subgroup->MakeNullParameter(PARAM_SRC_STR(0, chan),
+					B_MEDIA_RAW_AUDIO, StringForChannelType(buf,
+						out->GetOutputChannelType(chan)), B_GENERIC);
 				for (int src = 0; src < MAX_CHANNEL_TYPES; src++) {
 					subsubgroup = subgroup->MakeGroup("");
-					subsubgroup->MakeDiscreteParameter(PARAM_SRC_ENABLE(0, chan, src), B_MEDIA_RAW_AUDIO, "", B_ENABLE); 
-					subsubgroup->MakeContinuousParameter(PARAM_SRC_GAIN(0, chan, src), B_MEDIA_RAW_AUDIO,
-														 StringForChannelType(buf, src), B_GAIN, "%", 0.0, 100.0, 0.1);
+					subsubgroup->MakeDiscreteParameter(
+						PARAM_SRC_ENABLE(0, chan, src), B_MEDIA_RAW_AUDIO, "",
+						B_ENABLE);
+					subsubgroup->MakeContinuousParameter(
+						PARAM_SRC_GAIN(0, chan, src), B_MEDIA_RAW_AUDIO,
+						StringForChannelType(buf, src), B_GAIN, "%", 0.0,
+						100.0, 0.1);
 				}
 			}
 		}
 	}
-				
+
 	if (fCore->Settings()->AllowInputChannelRemapping()) {
 		top = web->MakeGroup("Input Mapping"); // top level group
 		inputchannels = top->MakeGroup("");
@@ -1692,7 +1769,7 @@ AudioMixer::UpdateParameterWeb()
 
 		for (int i = 0; (in = fCore->Input(i)); i++) {
 			group = inputchannels->MakeGroup("");
-			group->MakeNullParameter(PARAM_STR4(in->ID()), B_MEDIA_RAW_AUDIO, in->MediaInput().name, B_GENERIC); 
+			group->MakeNullParameter(PARAM_STR4(in->ID()), B_MEDIA_RAW_AUDIO, in->MediaInput().name, B_GENERIC);
 			group = group->MakeGroup("");
 
 			for (int chan = 0; chan < in->GetInputChannelCount(); chan++) {
@@ -1700,7 +1777,7 @@ AudioMixer::UpdateParameterWeb()
 				subgroup->MakeNullParameter(PARAM_DST_STR(in->ID(), chan), B_MEDIA_RAW_AUDIO,
 											StringForChannelType(buf, in->GetInputChannelType(chan)), B_GENERIC);
 				for (int dst = 0; dst < MAX_CHANNEL_TYPES; dst++) {
-					subgroup->MakeDiscreteParameter(PARAM_DST_ENABLE(in->ID(), chan, dst), B_MEDIA_RAW_AUDIO, StringForChannelType(buf, dst), B_ENABLE); 
+					subgroup->MakeDiscreteParameter(PARAM_DST_ENABLE(in->ID(), chan, dst), B_MEDIA_RAW_AUDIO, StringForChannelType(buf, dst), B_ENABLE);
 				}
 			}
 		}
@@ -1708,7 +1785,7 @@ AudioMixer::UpdateParameterWeb()
 
 	top = web->MakeGroup("Setup"); // top level group
 	group = top->MakeGroup("");
-	
+
 	group->MakeDiscreteParameter(PARAM_ETC(10), B_MEDIA_RAW_AUDIO, "Attenuate mixer output by 3dB (like BeOS R5)", B_ENABLE);
 	group->MakeDiscreteParameter(PARAM_ETC(20), B_MEDIA_RAW_AUDIO, "Use non linear gain sliders (like BeOS R5)", B_ENABLE);
 	group->MakeDiscreteParameter(PARAM_ETC(30), B_MEDIA_RAW_AUDIO, "Display balance control for stereo connections", B_ENABLE);
@@ -1737,9 +1814,11 @@ AudioMixer::UpdateParameterWeb()
 	SetParameterWeb(web);
 }
 
+
 #if USE_MEDIA_FORMAT_WORKAROUND
 static void
-raw_audio_format_specialize(media_raw_audio_format *format, const media_raw_audio_format *other)
+raw_audio_format_specialize(media_raw_audio_format *format,
+	const media_raw_audio_format *other)
 {
 	if (format->frame_rate == 0)
 		format->frame_rate = other->frame_rate;
@@ -1755,8 +1834,10 @@ raw_audio_format_specialize(media_raw_audio_format *format, const media_raw_audi
 		format->frame_rate = other->frame_rate;
 }
 
+
 static void
-multi_audio_info_specialize(media_multi_audio_info *format, const media_multi_audio_info *other)
+multi_audio_info_specialize(media_multi_audio_info *format,
+	const media_multi_audio_info *other)
 {
 	if (format->channel_mask == 0)
 		format->channel_mask = other->channel_mask;
@@ -1766,8 +1847,10 @@ multi_audio_info_specialize(media_multi_audio_info *format, const media_multi_au
 		format->matrix_mask = other->matrix_mask;
 }
 
+
 static void
-multi_audio_format_specialize(media_multi_audio_format *format, const media_multi_audio_format *other)
+multi_audio_format_specialize(media_multi_audio_format *format,
+	const media_multi_audio_format *other)
 {
 	raw_audio_format_specialize(format, other);
 	multi_audio_info_specialize(format, other);