media-add-ons/mixer/AudioMixer.cpp

/*
 * Copyright 2002 David Shipman,
 * Copyright 2003-2007 Marcus Overhagen
 * Copyright 2007 Haiku Inc. All rights reserved.
 * Distributed under the terms of the MIT License.
 */
#include "AudioMixer.h"
#include "MixerCore.h"
#include "MixerInput.h"
#include "MixerOutput.h"
#include "MixerUtils.h"

#include <Buffer.h>
#include <FindDirectory.h>
#include <math.h>
#include <MediaRoster.h>
#include <ParameterWeb.h>
#include <Path.h>
#include <RealtimeAlloc.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <TimeSource.h>

// the range of the gain sliders (in dB)
#define DB_MAX	18.0
#define DB_MIN	-60.0
// when using non linear sliders, we use a power function with
#define DB_EXPONENT_POSITIVE 1.4	// for dB values > 0
#define DB_EXPONENT_NEGATIVE 1.8	// for dB values < 0

#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)
#define GAIN_TO_PERCENT(gain)	((gain)  * 100.0)

// the id is encoded with 16 bits
// then chan and src (or dst) are encoded with 6 bits
// the unique number with 4 bits
// the PARAM_ETC etc is encoded with 26 bits
#define PARAM_SRC_ENABLE(id, chan, src)		(((id) << 16) | ((chan) << 10) | ((src) << 4) | 0x1)
#define PARAM_SRC_GAIN(id, chan, src)		(((id) << 16) | ((chan) << 10) | ((src) << 4) | 0x2)
#define PARAM_DST_ENABLE(id, chan, dst)		(((id) << 16) | ((chan) << 10) | ((dst) << 4) | 0x3)
#define PARAM_ETC(etc)						(((etc) << 16) | 0x4)
#define PARAM_SRC_STR(id, chan)				(((id) << 16) | ((chan) << 10) | 0x5)
#define PARAM_DST_STR(id, chan)				(((id) << 16) | ((chan) << 10) | 0x6)
#define PARAM_MUTE(id)						(((id) << 16) | 0x7)
#define PARAM_GAIN(id)						(((id) << 16) | 0x8)
#define PARAM_BALANCE(id)					(((id) << 16) | 0x9)
#define PARAM_STR1(id)						(((id) << 16) | ((1) << 10) | 0xa)
#define PARAM_STR2(id)						(((id) << 16) | ((2) << 10) | 0xa)
#define PARAM_STR3(id)						(((id) << 16) | ((3) << 10) | 0xa)
#define PARAM_STR4(id)						(((id) << 16) | ((4) << 10) | 0xa)
#define PARAM_STR5(id)						(((id) << 16) | ((5) << 10) | 0xa)
#define PARAM_STR6(id)						(((id) << 16) | ((6) << 10) | 0xa)
#define PARAM_STR7(id)						(((id) << 16) | ((7) << 10) | 0xa)

#define PARAM(id)							((id) >> 16)
#define ETC(id)								((id) >> 16)
#define PARAM_CHAN(id)						(((id) >> 10) & 0x3f)
#define PARAM_SRC(id)						(((id) >> 4) & 0x3f)
#define PARAM_DST(id)						(((id) >> 4) & 0x3f)
#define PARAM_IS_SRC_ENABLE(id)				(((id) & 0xf) == 0x1)
#define PARAM_IS_SRC_GAIN(id)				(((id) & 0xf) == 0x2)
#define PARAM_IS_DST_ENABLE(id)				(((id) & 0xf) == 0x3)
#define PARAM_IS_ETC(id)					(((id) & 0xf) == 0x4)
#define PARAM_IS_MUTE(id)					(((id) & 0xf) == 0x7)
#define PARAM_IS_GAIN(id)					(((id) & 0xf) == 0x8)
#define PARAM_IS_BALANCE(id)				(((id) & 0xf) == 0x9)


status_t
AudioMixer::GetParameterValue(int32 id, bigtime_t *last_change,
							  void *value, size_t *ioSize)
{
	TRACE("GetParameterValue: id 0x%08lx, ioSize %ld\n", id, *ioSize);
	int param = PARAM(id);
	fCore->Lock();
	if (PARAM_IS_ETC(id)) {
		switch (ETC(id)) {
			case 10:	// Attenuate mixer output by 3dB
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->AttenuateOutput();
				break;
			case 20:	// Use non linear gain sliders
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->NonLinearGainSlider();
				break;
			case 30:	// Display balance control for stereo connections
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->UseBalanceControl();
				break;
			case 40:	// Allow output channel remapping
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->AllowOutputChannelRemapping();
				break;
			case 50:	// Allow input channel remapping
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->AllowInputChannelRemapping();
				break;
			case 60:	// Input gain controls
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->InputGainControls();
				break;
			case 70:	// Resampling algorithm
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->ResamplingAlgorithm();
				break;
			case 80:	// Refuse output format changes
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->RefuseOutputFormatChange();
				break;
			case 90:	// Refuse input format changes
				*ioSize = sizeof(int32);
				static_cast<int32 *>(value)[0] = fCore->Settings()->RefuseInputFormatChange();
				break;
			default:
				ERROR("unhandled ETC 0x%08lx\n", id);
				break;
		}
	} else if (param == 0) {
		MixerOutput *output = fCore->Output();
		if (!output || (!PARAM_IS_MUTE(id) && !PARAM_IS_GAIN(id) && !PARAM_IS_SRC_ENABLE(id) && !PARAM_IS_SRC_GAIN(id) && !PARAM_IS_BALANCE(id)))
			goto err;
		if (PARAM_IS_MUTE(id)) {
			// output mute control
			if (*ioSize < sizeof(int32))
				goto err;
			*ioSize = sizeof(int32);
			static_cast<int32 *>(value)[0] = output->IsMuted();
		}
		if (PARAM_IS_GAIN(id)) {
			// output gain control
			if (fCore->Settings()->UseBalanceControl() && output->GetOutputChannelCount() == 2 && 1 /*channel mask is stereo */) {
				// single channel control + balance
				if (*ioSize < sizeof(float))
					goto err;
				*ioSize = sizeof(float);
				static_cast<float *>(value)[0] = GAIN_TO_DB((output->GetOutputChannelGain(0) + output->GetOutputChannelGain(1)) / 2);
			} else {
				// multi channel control
				if (*ioSize < output->GetOutputChannelCount() * sizeof(float))
					goto err;
				*ioSize = output->GetOutputChannelCount() * sizeof(float);
				for (int chan = 0; chan < output->GetOutputChannelCount(); chan++)
					static_cast<float *>(value)[chan] = GAIN_TO_DB(output->GetOutputChannelGain(chan));
			}
		}
		if (PARAM_IS_BALANCE(id)) {
			float l = output->GetOutputChannelGain(0);
			float r = output->GetOutputChannelGain(1);
			float v = r / (l+r);
			TRACE("balance l %1.3f, r %1.3f, v %1.3f\n",l,r,v);
			if (*ioSize < sizeof(float))
				goto err;
			*ioSize = sizeof(float);
			static_cast<float *>(value)[0] = v * 100;
		}
		if (PARAM_IS_SRC_ENABLE(id)) {
			if (*ioSize < sizeof(int32))
				goto err;
			*ioSize = sizeof(int32);
			static_cast<int32 *>(value)[0] = output->HasOutputChannelSource(PARAM_CHAN(id), PARAM_SRC(id));
		}
		if (PARAM_IS_SRC_GAIN(id)) {
			if (*ioSize < sizeof(float))
				goto err;
			*ioSize = sizeof(float);
			static_cast<float *>(value)[0] = GAIN_TO_PERCENT(output->GetOutputChannelSourceGain(PARAM_CHAN(id), PARAM_SRC(id)));
		}
	} else {
		MixerInput *input;
		for (int i = 0; (input = fCore->Input(i)); i++)
			if (input->ID() == param)
				break;
		if (!input || (!PARAM_IS_MUTE(id) && !PARAM_IS_GAIN(id) && !PARAM_IS_DST_ENABLE(id) && !PARAM_IS_BALANCE(id)))
			goto err;
		if (PARAM_IS_MUTE(id)) {
			// input mute control
			if (*ioSize < sizeof(int32))
				goto err;
			*ioSize = sizeof(int32);
			static_cast<int32 *>(value)[0] = !input->IsEnabled();
		}
		if (PARAM_IS_GAIN(id)) {
			// input gain control
			if (fCore->Settings()->InputGainControls() == 0) {
				// Physical input channels
				if (fCore->Settings()->UseBalanceControl() && input->GetInputChannelCount() == 2 && 1 /*channel mask is stereo */) {
					// single channel control + balance
					if (*ioSize < sizeof(float))
						goto err;
					*ioSize = sizeof(float);
					static_cast<float *>(value)[0] = GAIN_TO_DB((input->GetInputChannelGain(0) + input->GetInputChannelGain(1)) / 2);
				} else {
					// multi channel control
					if (*ioSize < input->GetInputChannelCount() * sizeof(float))
						goto err;
					*ioSize = input->GetInputChannelCount() * sizeof(float);
					for (int chan = 0; chan < input->GetInputChannelCount(); chan++)
						static_cast<float *>(value)[chan] = GAIN_TO_DB(input->GetInputChannelGain(chan));
				}
			} else {
				// Virtual output channels
				if (fCore->Settings()->UseBalanceControl() && input->GetMixerChannelCount() == 2 && 1 /*channel mask is stereo */) {
					// single channel control + balance
					if (*ioSize < sizeof(float))
						goto err;
					*ioSize = sizeof(float);
					static_cast<float *>(value)[0] = GAIN_TO_DB((input->GetMixerChannelGain(0) + input->GetMixerChannelGain(1)) / 2);
				} else {
					// multi channel control
					if (*ioSize < input->GetMixerChannelCount() * sizeof(float))
						goto err;
					*ioSize = input->GetMixerChannelCount() * sizeof(float);
					for (int chan = 0; chan < input->GetMixerChannelCount(); chan++)
						static_cast<float *>(value)[chan] = GAIN_TO_DB(input->GetMixerChannelGain(chan));
				}
			}
		}
		if (PARAM_IS_BALANCE(id)) {
			if (fCore->Settings()->InputGainControls() == 0) {
				// Physical input channels
				float l = input->GetInputChannelGain(0);
				float r = input->GetInputChannelGain(1);
				float v = r / (l+r);
				TRACE("balance l %1.3f, r %1.3f, v %1.3f\n",l,r,v);
				if (*ioSize < sizeof(float))
					goto err;
				*ioSize = sizeof(float);
				static_cast<float *>(value)[0] = v * 100;
			} else {
				// Virtual output channels
				float l = input->GetMixerChannelGain(0);
				float r = input->GetMixerChannelGain(1);
				float v = r / (l+r);
				TRACE("balance l %1.3f, r %1.3f, v %1.3f\n",l,r,v);
				if (*ioSize < sizeof(float))
					goto err;
				*ioSize = sizeof(float);
				static_cast<float *>(value)[0] = v * 100;
			}
		}
		if (PARAM_IS_DST_ENABLE(id)) {
			if (*ioSize < sizeof(int32))
				goto err;
			*ioSize = sizeof(int32);
			static_cast<int32 *>(value)[0] = input->HasInputChannelDestination(PARAM_CHAN(id), PARAM_DST(id));
		}
	}
	*last_change = TimeSource()->Now(); // XXX we could do better
	fCore->Unlock();
	return B_OK;
err:
	fCore->Unlock();
	return B_ERROR;
}

void
AudioMixer::SetParameterValue(int32 id, bigtime_t when,
							  const void *value, size_t size)
{
	TRACE("SetParameterValue: id 0x%08lx, size %ld\n", id, size);
	bool update = false;
	int param = PARAM(id);
	fCore->Lock();
	if (PARAM_IS_ETC(id)) {
		switch (ETC(id)) {
			case 10:	// Attenuate mixer output by 3dB
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetAttenuateOutput(static_cast<const int32 *>(value)[0]);
				// this value is special (see MixerCore.h) and we need to notify the core
				fCore->SetOutputAttenuation((static_cast<const int32 *>(value)[0]) ? 0.708 : 1.0);
				break;
			case 20:	// Use non linear gain sliders
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetNonLinearGainSlider(static_cast<const int32 *>(value)[0]);
				update = true; // XXX should use BroadcastChangedParameter()
				break;
			case 30:	// Display balance control for stereo connections
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetUseBalanceControl(static_cast<const int32 *>(value)[0]);
				update = true;
				break;
			case 40:	// Allow output channel remapping
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetAllowOutputChannelRemapping(static_cast<const int32 *>(value)[0]);
				update = true;
				break;
			case 50:	// Allow input channel remapping
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetAllowInputChannelRemapping(static_cast<const int32 *>(value)[0]);
				update = true;
				break;
			case 60:	// Input gain controls represent
						// (0, "Physical input channels")
						// (1, "Virtual output channels")
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetInputGainControls(static_cast<const int32 *>(value)[0]);
				update = true; // XXX should use BroadcastChangedParameter()
				break;
			case 70:	// Resampling algorithm
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetResamplingAlgorithm(static_cast<const int32 *>(value)[0]);
				// XXX tell the core to change the algorithm
				break;
			case 80:	// Refuse output format changes
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetRefuseOutputFormatChange(static_cast<const int32 *>(value)[0]);
				break;
			case 90:	// Refuse input format changes
				if (size != sizeof(int32))
					goto err;
				fCore->Settings()->SetRefuseInputFormatChange(static_cast<const int32 *>(value)[0]);
				break;
			default:
				ERROR("unhandled ETC 0x%08lx\n", id);
				break;
		}
	} else if (param == 0) {
		MixerOutput *output = fCore->Output();
		if (!output || (!PARAM_IS_MUTE(id) && !PARAM_IS_GAIN(id) && !PARAM_IS_SRC_ENABLE(id) && !PARAM_IS_SRC_GAIN(id) && !PARAM_IS_BALANCE(id)))
			goto err;
		if (PARAM_IS_MUTE(id)) {
			// output mute control
			if (size != sizeof(int32))
				goto err;
			output->SetMuted(static_cast<const int32 *>(value)[0]);
		}
		if (PARAM_IS_GAIN(id)) {
			// output gain control
			if (fCore->Settings()->UseBalanceControl() && output->GetOutputChannelCount() == 2 && 1 /*channel mask is stereo */) {
				// single channel control + balance
				float l = output->GetOutputChannelGain(0);
				float r = output->GetOutputChannelGain(1);
				float m = (l + r) / 2;	// master volume
				float v = DB_TO_GAIN(static_cast<const float *>(value)[0]);
				float f = v / m;		// factor for both channels
				TRACE("gain set l %1.3f, r %1.3f, m %1.3f, v %1.3f, f %1.3f\n",l,r,m,v,f);
				output->SetOutputChannelGain(0, output->GetOutputChannelGain(0) * f);
				output->SetOutputChannelGain(1, output->GetOutputChannelGain(1) * f);
			} else {
				// multi channel control
				if (size < output->GetOutputChannelCount() * sizeof(float))
					goto err;
				for (int chan = 0; chan < output->GetOutputChannelCount(); chan++)
					output->SetOutputChannelGain(chan, DB_TO_GAIN(static_cast<const float *>(value)[chan]));
			}
		}
		if (PARAM_IS_BALANCE(id)) {
			float l = output->GetOutputChannelGain(0);
			float r = output->GetOutputChannelGain(1);
			float m = (l + r) / 2;	// master volume
			float v = static_cast<const float *>(value)[0] / 100; // current balance value
			float fl = 2 * (1 - v);	// left channel factor of master volume
			float fr = 2 * v;		// right channel factor of master volume
			TRACE("balance set l %1.3f, r %1.3f, m %1.3f, v %1.3f, fl %1.3f, fr %1.3f\n",l,r,m,v,fl,fr);
			output->SetOutputChannelGain(0, m * fl);
			output->SetOutputChannelGain(1, m * fr);
		}
		if (PARAM_IS_SRC_ENABLE(id)) {
			if (size != sizeof(int32))
				goto err;
			if (static_cast<const int32 *>(value)[0]) {
				output->AddOutputChannelSource(PARAM_CHAN(id), PARAM_SRC(id));
			} else {
				output->RemoveOutputChannelSource(PARAM_CHAN(id), PARAM_SRC(id));
			}
		}
		if (PARAM_IS_SRC_GAIN(id)) {
			if (size != sizeof(float))
				goto err;
			output->SetOutputChannelSourceGain(PARAM_CHAN(id), PARAM_SRC(id), PERCENT_TO_GAIN(static_cast<const float *>(value)[0]));
		}
		fCore->Settings()->SaveConnectionSettings(output);
	} else {
		MixerInput *input;
		for (int i = 0; (input = fCore->Input(i)); i++)
			if (input->ID() == param)
				break;
		if (!input || (!PARAM_IS_MUTE(id) && !PARAM_IS_GAIN(id) && !PARAM_IS_DST_ENABLE(id) && !PARAM_IS_BALANCE(id)))
			goto err;
		if (PARAM_IS_MUTE(id)) {
			// input mute control
			if (size != sizeof(int32))
				goto err;
			input->SetEnabled(!static_cast<const int32 *>(value)[0]);
		}
		if (PARAM_IS_GAIN(id)) {
			// input gain control
			if (fCore->Settings()->InputGainControls() == 0) {
				// Physical input channels
				if (fCore->Settings()->UseBalanceControl() && input->GetInputChannelCount() == 2 && 1 /*channel mask is stereo */) {
					// single channel control + balance
					float l = input->GetInputChannelGain(0);
					float r = input->GetInputChannelGain(1);
					float m = (l + r) / 2;	// master volume
					float v = DB_TO_GAIN(static_cast<const float *>(value)[0]);
					float f = v / m;		// factor for both channels
					TRACE("gain set l %1.3f, r %1.3f, m %1.3f, v %1.3f, f %1.3f\n",l,r,m,v,f);
					input->SetInputChannelGain(0, input->GetInputChannelGain(0) * f);
					input->SetInputChannelGain(1, input->GetInputChannelGain(1) * f);
				} else {
					// multi channel control
					if (size < input->GetInputChannelCount() * sizeof(float))
						goto err;
					for (int chan = 0; chan < input->GetInputChannelCount(); chan++)
						input->SetInputChannelGain(chan, DB_TO_GAIN(static_cast<const float *>(value)[chan]));
				}
			} else {
				// Virtual output channels
				if (fCore->Settings()->UseBalanceControl() && input->GetMixerChannelCount() == 2 && 1 /*channel mask is stereo */) {
					// single channel control + balance
					float l = input->GetMixerChannelGain(0);
					float r = input->GetMixerChannelGain(1);
					float m = (l + r) / 2;	// master volume
					float v = DB_TO_GAIN(static_cast<const float *>(value)[0]);
					float f = v / m;		// factor for both channels
					TRACE("gain set l %1.3f, r %1.3f, m %1.3f, v %1.3f, f %1.3f\n",l,r,m,v,f);
					input->SetMixerChannelGain(0, input->GetMixerChannelGain(0) * f);
					input->SetMixerChannelGain(1, input->GetMixerChannelGain(1) * f);
				} else {
					// multi channel control
					if (size < input->GetMixerChannelCount() * sizeof(float))
						goto err;
					for (int chan = 0; chan < input->GetMixerChannelCount(); chan++)
						input->SetMixerChannelGain(chan, DB_TO_GAIN(static_cast<const float *>(value)[chan]));
				}
			}
		}
		if (PARAM_IS_BALANCE(id)) {
			if (fCore->Settings()->InputGainControls() == 0) {
				// Physical input channels
				float l = input->GetInputChannelGain(0);
				float r = input->GetInputChannelGain(1);
				float m = (l + r) / 2;	// master volume
				float v = static_cast<const float *>(value)[0] / 100; // current balance value
				float fl = 2 * (1 - v);	// left channel factor of master volume
				float fr = 2 * v;		// right channel factor of master volume
				TRACE("balance set l %1.3f, r %1.3f, m %1.3f, v %1.3f, fl %1.3f, fr %1.3f\n",l,r,m,v,fl,fr);
				input->SetInputChannelGain(0, m * fl);
				input->SetInputChannelGain(1, m * fr);
			} else {
				// Virtual output channels
				float l = input->GetMixerChannelGain(0);
				float r = input->GetMixerChannelGain(1);
				float m = (l + r) / 2;	// master volume
				float v = static_cast<const float *>(value)[0] / 100; // current balance value
				float fl = 2 * (1 - v);	// left channel factor of master volume
				float fr = 2 * v;		// right channel factor of master volume
				TRACE("balance set l %1.3f, r %1.3f, m %1.3f, v %1.3f, fl %1.3f, fr %1.3f\n",l,r,m,v,fl,fr);
				input->SetMixerChannelGain(0, m * fl);
				input->SetMixerChannelGain(1, m * fr);
			}
		}
		if (PARAM_IS_DST_ENABLE(id)) {
			if (size != sizeof(int32))
				goto err;
			if (static_cast<const int32 *>(value)[0]) {
				int oldchan = input->GetInputChannelForDestination(PARAM_DST(id));
				if (oldchan != -1) {
					input->RemoveInputChannelDestination(oldchan, PARAM_DST(id));
					int32 null = 0;
					BroadcastNewParameterValue(when, PARAM_DST_ENABLE(PARAM(id), oldchan, PARAM_DST(id)), &null, sizeof(null));
				}
				input->AddInputChannelDestination(PARAM_CHAN(id), PARAM_DST(id));
			} else {
				input->RemoveInputChannelDestination(PARAM_CHAN(id), PARAM_DST(id));
			}
			// XXX 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.
			// BroadcastChangedParameter() should be correct, but doesn't work
			BroadcastChangedParameter(PARAM_GAIN(PARAM(id)));
			// We trigger a complete ParameterWeb update as workaround
			// but it will change the focus from tab 3 to tab 1
			update = true;
		}
		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();
	BParameterGroup *top;
	BParameterGroup *outputchannels;
	BParameterGroup *inputchannels;
	BParameterGroup *group;
	BParameterGroup *subgroup;
	BParameterGroup *subsubgroup;
	BDiscreteParameter *dp;
	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);
	if (!out) {
		group->MakeNullParameter(PARAM_STR2(0), B_MEDIA_RAW_AUDIO, "not connected", B_GENERIC);
	} else {
		group->MakeNullParameter(PARAM_STR2(0), B_MEDIA_RAW_AUDIO, StringForFormat(buf, out), B_GENERIC);
		group->MakeDiscreteParameter(PARAM_MUTE(0), B_MEDIA_RAW_AUDIO, "Mute", B_MUTE);
		if (fCore->Settings()->UseBalanceControl() && out->GetOutputChannelCount() == 2 && 1 /*channel mask is stereo */) {
			// single channel control + balance
			group->MakeContinuousParameter(PARAM_GAIN(0), B_MEDIA_RAW_AUDIO, "Gain", B_MASTER_GAIN, "dB", DB_MIN, DB_MAX, 0.1);
			group->MakeContinuousParameter(PARAM_BALANCE(0), B_MEDIA_RAW_AUDIO, "", B_BALANCE, "", 0, 100, 1);
		} 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());
		}
		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_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,
		//     as you don't know what channel each slider controls. Tooltips might help...
		if (fCore->Settings()->InputGainControls() == 0) {
			// Physical input channels
			if (fCore->Settings()->UseBalanceControl() && in->GetInputChannelCount() == 2 && 1 /*channel mask is stereo */) {
				// single channel control + balance
				group->MakeContinuousParameter(PARAM_GAIN(in->ID()), B_MEDIA_RAW_AUDIO, "Gain", B_GAIN, "dB", DB_MIN, DB_MAX, 0.1);
				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)
											   ->SetChannelCount(in->GetInputChannelCount());
			}
		} else {
			// Virtual output channels
			if (fCore->Settings()->UseBalanceControl() && in->GetMixerChannelCount() == 2 && 1 /*channel mask is stereo */) {
				// single channel control + balance
				group->MakeContinuousParameter(PARAM_GAIN(in->ID()), B_MEDIA_RAW_AUDIO, "Gain", B_GAIN, "dB", DB_MIN, DB_MAX, 0.1);
				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)
											   ->SetChannelCount(in->GetMixerChannelCount());
			}
		}
		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 = 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);
				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);
				}
			}
		}
	}

	if (fCore->Settings()->AllowInputChannelRemapping()) {
		top = web->MakeGroup("Input Mapping"); // top level group
		inputchannels = top->MakeGroup("");
		inputchannels->MakeNullParameter(PARAM_STR7(0), B_MEDIA_RAW_AUDIO, "Input Channel Destinations", B_GENERIC);

		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 = group->MakeGroup("");

			for (int chan = 0; chan < in->GetInputChannelCount(); chan++) {
				subgroup = group->MakeGroup("");
				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);
				}
			}
		}
	}

	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);

	group->MakeDiscreteParameter(PARAM_ETC(40), B_MEDIA_RAW_AUDIO, "Allow output channel remapping", B_ENABLE);
	group->MakeDiscreteParameter(PARAM_ETC(50), B_MEDIA_RAW_AUDIO, "Allow input channel remapping", B_ENABLE);

	dp = group->MakeDiscreteParameter(PARAM_ETC(60), B_MEDIA_RAW_AUDIO, "Input gain controls represent", B_INPUT_MUX);
	dp->AddItem(0, "Physical input channels");
	dp->AddItem(1, "Virtual output channels");

	dp = group->MakeDiscreteParameter(PARAM_ETC(70), B_MEDIA_RAW_AUDIO, "Resampling algorithm", B_INPUT_MUX);
	dp->AddItem(0, "Drop/repeat samples");

	// Note: The following code is outcommented on purpose
	// and is about to be modified at a later point
	/*
	dp->AddItem(1, "Drop/repeat samples (template based)");
	dp->AddItem(2, "Linear interpolation");
	dp->AddItem(3, "17th order filtering");
	*/
	group->MakeDiscreteParameter(PARAM_ETC(80), B_MEDIA_RAW_AUDIO, "Refuse output format changes", B_ENABLE);
	group->MakeDiscreteParameter(PARAM_ETC(90), B_MEDIA_RAW_AUDIO, "Refuse input format changes", B_ENABLE);

	fCore->Unlock();
	SetParameterWeb(web);
}

#if USE_MEDIA_FORMAT_WORKAROUND
static void
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;
	if (format->channel_count == 0)
		format->channel_count = other->channel_count;
	if (format->format == 0)
		format->format = other->format;
	if (format->byte_order == 0)
		format->byte_order = other->byte_order;
	if (format->buffer_size == 0)
		format->buffer_size = other->buffer_size;
	if (format->frame_rate == 0)
		format->frame_rate = other->frame_rate;
}

static void
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;
	if (format->valid_bits == 0)
		format->valid_bits = other->valid_bits;
	if (format->matrix_mask == 0)
		format->matrix_mask = other->matrix_mask;
}

static void
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);
}
#endif