VoIP mixing and decoding does now happen in the batch processing thread instead of immediately. Additionally, the system now supports filter plugins such as codecs.

git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@120 09dcff7a-b715-0410-9601-b79a96267cd0

18 files changed, 925 insertions, 40 deletions

diff --git a/orkaudio/AudioTape.cpp b/orkaudio/AudioTape.cpp
index fd01b2c..c7ea945 100644
--- a/orkaudio/AudioTape.cpp
+++ b/orkaudio/AudioTape.cpp
@@ -16,9 +16,10 @@
 #include "ace/OS_NS_time.h"
 #include "Utils.h"
 #include "ThreadSafeQueue.h"
-#include "audiofile/PcmFile.h"
 #include "LogManager.h"
+#include "audiofile/PcmFile.h"
 #include "audiofile/LibSndFileFile.h"
+#include "audiofile/MediaChunkFile.h"
 #include "messages/TapeMsg.h"
 
 AudioTapeDescription::AudioTapeDescription()
@@ -69,6 +70,7 @@ AudioTape::AudioTape(CStdString &portId)
 	m_duration = 0;
 	m_direction = CaptureEvent::DirUnkn;
 	m_shouldStop = false;
+	m_readyForBatchProcessing = false;
 
 	GenerateFilePathAndIdentifier();
 }
@@ -77,6 +79,7 @@ AudioTape::AudioTape(CStdString &portId)
 void AudioTape::AddAudioChunk(AudioChunkRef chunkRef, bool remote)
 {
 	// Add the chunk to the local queue
+	if(m_state == StateCreated || m_state == StateActive)
 	{
 		MutexSentinel sentinel(m_mutex);
 		if(remote)
@@ -119,26 +122,30 @@ void AudioTape::Write()
 				{
 					m_state = StateActive;
 
-					switch(chunkRef->m_encoding)
+					switch(chunkRef->GetEncoding())
 					{
-					case AudioChunk::PcmAudio:
-						m_audioFileRef.reset(new PcmFile);
-						break;
-					case AudioChunk::UlawAudio:
-						m_audioFileRef.reset(new LibSndFileFile(SF_FORMAT_ULAW | SF_FORMAT_WAV));
-						break;
-					case AudioChunk::AlawAudio:
-						m_audioFileRef.reset(new LibSndFileFile(SF_FORMAT_ALAW | SF_FORMAT_WAV));
-						break;
+					case PcmAudio:
+						//m_audioFileRef.reset(new PcmFile);
+						//break;
+					case UlawAudio:
+						//m_audioFileRef.reset(new LibSndFileFile(SF_FORMAT_ULAW | SF_FORMAT_WAV));
+						//break;
+					case AlawAudio:
+						//m_audioFileRef.reset(new LibSndFileFile(SF_FORMAT_ALAW | SF_FORMAT_WAV));
+						//break;
 					default:
-						LOG4CXX_ERROR(LOG.portLog, "#" + m_portId + ": received unsupported audio encoding from capture plugin:" + FileFormatToString(chunkRef->m_encoding));
-						m_state = StateError;
+						//LOG4CXX_ERROR(LOG.portLog, "#" + m_portId + ": received unsupported audio encoding from capture plugin:" + FileFormatToString(chunkRef->GetEncoding()));
+						//m_state = StateError;						
+
+						// ###########
+						// All other encodings: output as a media chunk file
+						m_audioFileRef.reset(new MediaChunkFile());
 					}
 					if (m_state == StateActive)
 					{
 						// A file format was successfully added to the tape, open it
 						CStdString file = CONFIG.m_audioOutputPath + "/" + m_filePath + m_fileIdentifier;
-						m_audioFileRef->Open(file, AudioFile::WRITE, false, chunkRef->m_sampleRate);
+						m_audioFileRef->Open(file, AudioFile::WRITE, false, chunkRef->GetSampleRate());
 
 						// determine what final extension the file will have after optional compression
 						if(CONFIG.m_storageAudioFormat == FfNative)
@@ -172,18 +179,17 @@ void AudioTape::Write()
 		}
 	}
 
-	if (m_shouldStop)
+	if ( (m_shouldStop && m_state != StateStopped) || m_state == StateError)
 	{
 		m_state = StateStopped;
-	}
-
-	if (m_state == StateStopped || m_state == StateError)
-	{
 		if(m_audioFileRef.get())
 		{
 			m_audioFileRef->Close();
+			m_readyForBatchProcessing = true;
 		}
 	}
+
+
 }
 
 void AudioTape::SetShouldStop()
@@ -317,6 +323,17 @@ AudioFileRef AudioTape::GetAudioFileRef()
 	return m_audioFileRef;
 }
 
+bool AudioTape::IsReadyForBatchProcessing()
+{
+	if(m_readyForBatchProcessing)
+	{
+		m_readyForBatchProcessing = false;	// toggle to ensure not processed twice
+		return true;
+	}
+	return false;
+}
+
+
 //========================================
 // File format related methods
 
diff --git a/orkaudio/AudioTape.h b/orkaudio/AudioTape.h
index 7d40564..9870c46 100644
--- a/orkaudio/AudioTape.h
+++ b/orkaudio/AudioTape.h
@@ -90,6 +90,7 @@ public:
 	CStdString GetIdentifier();
 	CStdString GetPath();
 	AudioFileRef GetAudioFileRef();
+	bool IsReadyForBatchProcessing();
 
 	CStdString m_portId;
 	CStdString m_localParty;
@@ -116,6 +117,7 @@ private:
 	ACE_Thread_Mutex m_mutex;
 	StateEnum m_state;
 	bool m_shouldStop;
+	bool m_readyForBatchProcessing;
 };
 
 typedef boost::shared_ptr<AudioTape> AudioTapeRef;
diff --git a/orkaudio/BatchProcessing.cpp b/orkaudio/BatchProcessing.cpp
index 1a01ac5..90f2b7b 100644
--- a/orkaudio/BatchProcessing.cpp
+++ b/orkaudio/BatchProcessing.cpp
@@ -17,6 +17,7 @@
 #include "ace/OS_NS_unistd.h"
 #include "audiofile/LibSndFileFile.h"
 #include "Daemon.h"
+#include "Filter.h"
 
 BatchProcessing BatchProcessing::m_batchProcessingSingleton;
 
@@ -42,6 +43,8 @@ void BatchProcessing::AddAudioTape(AudioTapeRef audioTapeRef)
 
 void BatchProcessing::ThreadHandler(void *args)
 {
+	CStdString debug;
+
 	BatchProcessing* pBatchProcessing = BatchProcessing::GetInstance();
 	int threadId = 0;
 	{
@@ -55,6 +58,9 @@ void BatchProcessing::ThreadHandler(void *args)
 
 	for(;stop == false;)
 	{
+		AudioFileRef fileRef;
+		AudioFileRef outFileRef;
+
 		try
 		{
 			AudioTapeRef audioTapeRef = pBatchProcessing->m_audioTapeQueue.pop();
@@ -70,12 +76,11 @@ void BatchProcessing::ThreadHandler(void *args)
 				CStdString threadIdString = IntToString(threadId);
 				LOG4CXX_INFO(LOG.batchProcessingLog, CStdString("Th") + threadIdString + " processing: " + audioTapeRef->GetIdentifier());
 
-				AudioFileRef fileRef = audioTapeRef->GetAudioFileRef();
+				fileRef = audioTapeRef->GetAudioFileRef();
 				fileRef->MoveOrig();
 				fileRef->Open(AudioFile::READ);
 
 				AudioChunkRef chunkRef;
-				AudioFileRef outFileRef;
 
 				switch(CONFIG.m_storageAudioFormat)
 				{
@@ -95,14 +100,54 @@ void BatchProcessing::ThreadHandler(void *args)
 				CStdString file = CONFIG.m_audioOutputPath + "/" + audioTapeRef->GetPath() + audioTapeRef->GetIdentifier();
 				outFileRef->Open(file, AudioFile::WRITE, false, fileRef->GetSampleRate());
 
+				FilterRef filter;
+				FilterRef decoder1;
+				FilterRef decoder2;
+
+				bool firstChunk = true;
+				bool voIpSession = false;
+
 				while(fileRef->ReadChunkMono(chunkRef))
 				{
+					AudioChunkDetails details = *chunkRef->GetDetails();
+					if(firstChunk && details.m_rtpPayloadType != -1)
+					{
+						firstChunk = false;
+						CStdString filterName("VoIpMixer");
+						filter = FilterRegistry::instance()->GetNewFilter(filterName);
+						decoder1 = FilterRegistry::instance()->GetNewFilter(details.m_rtpPayloadType);
+						decoder2 = FilterRegistry::instance()->GetNewFilter(details.m_rtpPayloadType);
+						if(decoder1.get() == NULL || decoder2.get() == NULL)
+						{
+							debug.Format("BatchProcessing - Could not find decoder for RTP payload type:%u", chunkRef->GetDetails()->m_rtpPayloadType);
+							throw(debug);
+						}
+						voIpSession = true;
+					}
+					if(voIpSession)
+					{
+						if(details.m_channel == 2)
+						{
+							decoder2->AudioChunkIn(chunkRef);
+							decoder2->AudioChunkOut(chunkRef);
+						}
+						else
+						{
+							decoder1->AudioChunkIn(chunkRef);
+							decoder1->AudioChunkOut(chunkRef);
+						}
+						
+						filter->AudioChunkIn(chunkRef);
+						filter->AudioChunkOut(chunkRef);
+					}
 					outFileRef->WriteChunk(chunkRef);
 				}
 
+				fileRef->Close();
+				outFileRef->Close();
+
 				if(CONFIG.m_deleteNativeFile)
 				{
-					fileRef->Close();
 					fileRef->Delete();
 					CStdString threadIdString = IntToString(threadId);
 					LOG4CXX_INFO(LOG.batchProcessingLog, CStdString("Th") + threadIdString + " deleting native: " + audioTapeRef->GetIdentifier());
@@ -113,9 +158,10 @@ void BatchProcessing::ThreadHandler(void *args)
 		{
 			LOG4CXX_ERROR(LOG.batchProcessingLog, CStdString("BatchProcessing: ") + e);
 		}
-		catch(...)
-		{
-		}
+		//catch(...)
+		//{
+		//	LOG4CXX_ERROR(LOG.batchProcessingLog, CStdString("BatchProcessing: unknown exception"));
+		//}
 	}
 	LOG4CXX_INFO(LOG.batchProcessingLog, CStdString("Exiting thread #" + threadIdString));
 }
diff --git a/orkaudio/CapturePort.cpp b/orkaudio/CapturePort.cpp
index 52ca35c..fa5a4e1 100644
--- a/orkaudio/CapturePort.cpp
+++ b/orkaudio/CapturePort.cpp
@@ -74,7 +74,7 @@ void CapturePort::AddAudioChunk(AudioChunkRef chunkRef, bool remote)
 	}
 	else if (CONFIG.m_vad)
 	{
-		if(chunkRef->m_encoding == AudioChunk::PcmAudio)
+		if(chunkRef->GetEncoding() == PcmAudio)
 		{
 			if(m_vadUp)
 			{
diff --git a/orkaudio/ImmediateProcessing.cpp b/orkaudio/ImmediateProcessing.cpp
index 26f564e..1298117 100644
--- a/orkaudio/ImmediateProcessing.cpp
+++ b/orkaudio/ImmediateProcessing.cpp
@@ -10,6 +10,7 @@
  * Please refer to http://www.gnu.org/copyleft/gpl.html
  *
  */
+#pragma warning( disable: 4786 )
 
 #include "ImmediateProcessing.h"
 #include "LogManager.h"
@@ -58,7 +59,7 @@ void ImmediateProcessing::ThreadHandler(void *args)
 			
 				audioTapeRef->Write();
 
-				if (audioTapeRef->IsStoppedAndValid())
+				if (audioTapeRef->IsReadyForBatchProcessing())
 				{
 					// Forward to batch processing thread
 					BatchProcessing::GetInstance()->AddAudioTape(audioTapeRef);
diff --git a/orkaudio/OrkAudio.cpp b/orkaudio/OrkAudio.cpp
index 05ac4c1..daea814 100644
--- a/orkaudio/OrkAudio.cpp
+++ b/orkaudio/OrkAudio.cpp
@@ -16,6 +16,7 @@
 
 #include "MultiThreadedServer.h"
 #include "ace/Thread_Manager.h"
+#include "ace/DLL.h"
 #include "OrkAudio.h"
 #include "Utils.h"
 #include "messages/TapeMsg.h"
@@ -33,6 +34,8 @@
 #include "ObjectFactory.h"
 #include "CapturePluginProxy.h"
 #include "ace/OS_NS_arpa_inet.h"
+#include "AudioCapturePlugin.h"
+#include "Filter.h"
 
 
 static volatile bool serviceStop = false;
@@ -73,6 +76,35 @@ void MainThread()
 
 	ConfigManagerSingleton::instance()->Initialize();
 
+	// Register in-built filters
+	FilterRef filter(new AlawToPcmFilter());
+	FilterRegistry::instance()->RegisterFilter(filter);
+
+	// Load filter plugins  #####################
+	//CStdString pluginPath = "./filters/VoIpMixer/Debug/VoIpMixer.dll";
+	CStdString pluginPath = "./filters/VoIpMixer.dll";
+	ACE_DLL dll;
+	dll.open((PCSTR)pluginPath);
+	ACE_TCHAR* error = dll.error();
+	if(error)
+	{
+		LOG4CXX_ERROR(LOG.rootLog, CStdString("Failed to load the following plugin: ") + pluginPath);
+	}
+	else
+	{
+		// Ok, the dll has been successfully loaded
+		LOG4CXX_INFO(LOG.rootLog, CStdString("Loaded plugin: ") + pluginPath);
+
+		//void (*initfunction)(void);
+		InitializeFunction initfunction;
+		initfunction = (InitializeFunction)dll.symbol("Initialize");
+
+		if (initfunction)
+		{
+			initfunction();
+		}
+	}
+
 	if (!ACE_Thread_Manager::instance()->spawn(ACE_THR_FUNC(ImmediateProcessing::ThreadHandler)))
 	{
 		LOG4CXX_INFO(LOG.rootLog, CStdString("Failed to create immediate processing thread"));
diff --git a/orkaudio/OrkAudio.dsp b/orkaudio/OrkAudio.dsp
index d32442d..8e21d3f 100644
--- a/orkaudio/OrkAudio.dsp
+++ b/orkaudio/OrkAudio.dsp
@@ -150,6 +150,14 @@ SOURCE=.\AudioFile\LibSndFileFile.h
 # End Source File
 # Begin Source File
 
+SOURCE=.\audiofile\MediaChunkFile.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=.\audiofile\MediaChunkFile.h
+# End Source File
+# Begin Source File
+
 SOURCE=.\AudioFile\PcmFile.cpp
 # End Source File
 # Begin Source File
diff --git a/orkaudio/OrkAudio.dsw b/orkaudio/OrkAudio.dsw
index aa80c9f..625b1df 100644
--- a/orkaudio/OrkAudio.dsw
+++ b/orkaudio/OrkAudio.dsw
@@ -63,6 +63,18 @@ Package=<4>
 
 ###############################################################################
 
+Project: "VoIpMixer"=.\filters\VoIpMixer\VoIpMixer.dsp - Package Owner=<4>
+
+Package=<5>
+{{{
+}}}
+
+Package=<4>
+{{{
+}}}
+
+###############################################################################
+
 Global:
 
 Package=<5>
diff --git a/orkaudio/audiocaptureplugins/generator/Generator.cpp b/orkaudio/audiocaptureplugins/generator/Generator.cpp
index 8deb07a..d97d0e4 100644
--- a/orkaudio/audiocaptureplugins/generator/Generator.cpp
+++ b/orkaudio/audiocaptureplugins/generator/Generator.cpp
@@ -126,7 +126,9 @@ void Run()
 			// send audio buffer
 			AudioChunkRef chunkRef(new AudioChunk);
 			int sampleOffset = (elapsed % numChunks)*NUM_SAMPLES_PER_CHUNK;
-			chunkRef->SetBuffer(audioBuffer+sampleOffset, sizeof(short)*NUM_SAMPLES_PER_CHUNK, AudioChunk::PcmAudio);
+			AudioChunkDetails details;
+			details.m_encoding = PcmAudio;
+			chunkRef->SetBuffer(audioBuffer+sampleOffset, sizeof(short)*NUM_SAMPLES_PER_CHUNK, details);
 			g_audioChunkCallBack(chunkRef, portName);
 		}
 
diff --git a/orkaudio/audiocaptureplugins/sounddevice/SoundDevice.cpp b/orkaudio/audiocaptureplugins/sounddevice/SoundDevice.cpp
index 5326152..db791ad 100644
--- a/orkaudio/audiocaptureplugins/sounddevice/SoundDevice.cpp
+++ b/orkaudio/audiocaptureplugins/sounddevice/SoundDevice.cpp
@@ -65,12 +65,20 @@ int portAudioCallBack(void *inputBuffer, void *outputBuffer, unsigned long frame
 			leftBuffer[sampleId] = inputSamples[2*sampleId+1];
 		}
 		AudioChunkRef chunkRef(new AudioChunk);
-		chunkRef->SetBuffer(rightBuffer, sizeof(short)*framesPerBuffer, AudioChunk::PcmAudio, 0, 0, DLLCONFIG.m_sampleRate);
+		AudioChunkDetails rightDetails;
+		rightDetails.m_encoding = PcmAudio;
+		rightDetails.m_sampleRate = DLLCONFIG.m_sampleRate;
+		rightDetails.m_channel = 1;
+		chunkRef->SetBuffer(rightBuffer, sizeof(short)*framesPerBuffer, rightDetails);
 		portName.Format("port%d-%d", device->deviceID, 1);
 		g_audioChunkCallBack(chunkRef, portName);
 
 		chunkRef.reset(new AudioChunk);
-		chunkRef->SetBuffer(leftBuffer, sizeof(short)*framesPerBuffer, AudioChunk::PcmAudio, 0, 0, DLLCONFIG.m_sampleRate);
+		AudioChunkDetails leftDetails;
+		leftDetails.m_encoding = PcmAudio;
+		leftDetails.m_sampleRate = DLLCONFIG.m_sampleRate;
+		leftDetails.m_channel = 1;
+		chunkRef->SetBuffer(leftBuffer, sizeof(short)*framesPerBuffer, leftDetails);
 		portName.Format("port%d-%d", device->deviceID, 2);
 		g_audioChunkCallBack(chunkRef, portName);
 
@@ -81,7 +89,11 @@ int portAudioCallBack(void *inputBuffer, void *outputBuffer, unsigned long frame
 	{
 		// mono
 		AudioChunkRef chunkRef(new AudioChunk);
-		chunkRef->SetBuffer(inputSamples, sizeof(short)*framesPerBuffer, AudioChunk::PcmAudio, 0, 0, DLLCONFIG.m_sampleRate);
+		AudioChunkDetails details;
+		details.m_encoding = PcmAudio;
+		details.m_sampleRate = DLLCONFIG.m_sampleRate;
+		details.m_channel = 1;
+		chunkRef->SetBuffer(inputSamples, sizeof(short)*framesPerBuffer, details);
 		portName.Format("port%d", device->deviceID);
 		g_audioChunkCallBack(chunkRef, portName);
 	}
diff --git a/orkaudio/audiocaptureplugins/voip/Rtp.cpp b/orkaudio/audiocaptureplugins/voip/Rtp.cpp
index 23c236e..49d0815 100644
--- a/orkaudio/audiocaptureplugins/voip/Rtp.cpp
+++ b/orkaudio/audiocaptureplugins/voip/Rtp.cpp
@@ -36,6 +36,7 @@ void RtpPacketInfo::ToString(CStdString& string)
 	string.Format("%s,%d %s,%d seq:%u ts:%u len:%d type:%x", sourceIp, m_sourcePort, destIp, m_destPort, m_seqNum, m_timestamp, m_payloadSize, m_payloadType);
 }
 
+//==============================================================================
 
 RtpRingBuffer::RtpRingBuffer()
 {
@@ -199,7 +200,9 @@ void RtpRingBuffer::CreateShipment(size_t silenceSize)
 	size_t shortSize = stopPtr-m_readPtr;
 	size_t byteSize = shortSize*2;
 	AudioChunkRef chunk(new AudioChunk());
-	chunk->SetBuffer((void*)m_readPtr, byteSize, AudioChunk::PcmAudio);
+	AudioChunkDetails details;
+	details.m_encoding = PcmAudio;
+	chunk->SetBuffer((void*)m_readPtr, byteSize, details);
 	g_audioChunkCallBack(chunk, m_capturePort);
 	m_shippedSamples += shortSize;
 	m_readPtr = CircularPointerAddOffset(m_readPtr ,shortSize);
@@ -216,7 +219,9 @@ void RtpRingBuffer::CreateShipment(size_t silenceSize)
 		shortSize = wrappedStopPtr - m_buffer;
 		byteSize = shortSize*2;
 		chunk.reset(new AudioChunk());
-		chunk->SetBuffer((void*)m_buffer, byteSize, AudioChunk::PcmAudio);
+		AudioChunkDetails details;
+		details.m_encoding = PcmAudio;
+		chunk->SetBuffer((void*)m_buffer, byteSize, details);
 		g_audioChunkCallBack(chunk, m_capturePort);
 		m_shippedSamples += shortSize;
 		m_readPtr = CircularPointerAddOffset(m_readPtr ,shortSize);
@@ -233,7 +238,9 @@ void RtpRingBuffer::CreateShipment(size_t silenceSize)
 		if (silenceBuffer)
 		{
 			AudioChunkRef chunk(new AudioChunk());
-			chunk->SetBuffer((void*)silenceBuffer, byteSize, AudioChunk::PcmAudio);
+			AudioChunkDetails details;
+			details.m_encoding = PcmAudio;
+			chunk->SetBuffer((void*)silenceBuffer, byteSize, details);
 			g_audioChunkCallBack(chunk, m_capturePort);
 			m_shippedSamples += silenceSize;
 			m_readPtr = CircularPointerAddOffset(m_readPtr ,silenceSize);
diff --git a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
index 9065eb4..2cac30c 100644
--- a/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
+++ b/orkaudio/audiocaptureplugins/voip/RtpSession.cpp
@@ -237,6 +237,7 @@ void RtpSession::ReportMetadata()
 void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 {
 	CStdString logMsg;
+	unsigned char channel = 0;
 
 	if(m_lastRtpPacket.get() == NULL)
 	{
@@ -262,6 +263,7 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 	{
 		// First RTP packet for side 1
 		m_lastRtpPacketSide1 = rtpPacket;
+		channel = 1;
 		if(m_log->isInfoEnabled())
 		{
 			rtpPacket->ToString(logMsg);
@@ -274,6 +276,7 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 		if(rtpPacket->m_sourceIp.s_addr == m_lastRtpPacketSide1->m_sourceIp.s_addr)
 		{
 			m_lastRtpPacketSide1 = rtpPacket;
+			channel = 1;
 		}
 		else
 		{
@@ -288,6 +291,7 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 				}
 			}
 			m_lastRtpPacketSide2 = rtpPacket;
+			channel = 2;
 		}
 	}
 
@@ -333,7 +337,19 @@ void RtpSession::AddRtpPacket(RtpPacketInfoRef& rtpPacket)
 
 	if(m_started)
 	{
-		m_rtpRingBuffer.AddRtpPacket(rtpPacket);
+		AudioChunkDetails details;
+		details.m_arrivalTimestamp = rtpPacket->m_arrivalTimestamp;
+		details.m_numBytes = rtpPacket->m_payloadSize;
+		details.m_timestamp = rtpPacket->m_timestamp;
+		details.m_rtpPayloadType = rtpPacket->m_payloadType;
+		details.m_sequenceNumber = rtpPacket->m_seqNum;
+		details.m_channel = channel;
+		details.m_encoding = AlawAudio;
+		AudioChunkRef chunk(new AudioChunk());
+		chunk->SetBuffer(rtpPacket->m_payload, rtpPacket->m_payloadSize, details);
+		g_audioChunkCallBack(chunk, m_capturePort);	// ##### after
+		//m_rtpRingBuffer.AddRtpPacket(rtpPacket);	// ##### before
+
 		m_lastUpdated = rtpPacket->m_arrivalTimestamp;
 	}
 }
diff --git a/orkaudio/audiofile/LibSndFileFile.cpp b/orkaudio/audiofile/LibSndFileFile.cpp
index fbbf3cb..0589466 100644
--- a/orkaudio/audiofile/LibSndFileFile.cpp
+++ b/orkaudio/audiofile/LibSndFileFile.cpp
@@ -68,10 +68,18 @@ void LibSndFileFile::Open(CStdString& filename, fileOpenModeEnum mode, bool ster
 
 void LibSndFileFile::WriteChunk(AudioChunkRef chunkRef)
 {
-	if (m_pFile)
+	if(chunkRef.get() == NULL)
+	{
+		return;
+	}
+	if(chunkRef->GetDetails()->m_numBytes == 0)
 	{
+		return;
+	}
 
-		if( chunkRef->m_encoding == AudioChunk::AlawAudio ||  chunkRef->m_encoding == AudioChunk::UlawAudio)
+	if (m_pFile)
+	{
+		if( chunkRef->GetEncoding() == AlawAudio ||  chunkRef->GetEncoding() == UlawAudio)
 		{
 			if(sf_write_raw(m_pFile, chunkRef->m_pBuffer, chunkRef->GetNumSamples()) != chunkRef->GetNumSamples())
 			{
@@ -79,7 +87,7 @@ void LibSndFileFile::WriteChunk(AudioChunkRef chunkRef)
 				throw(CStdString("sf_write_raw failed, audio file " + m_filename + " could not be written after " + numChunksWrittenString + " chunks written"));
 			}
 		}
-		else if (chunkRef->m_encoding == AudioChunk::PcmAudio)
+		else if (chunkRef->GetEncoding() == PcmAudio)
 		{
 			if(sf_write_short(m_pFile, (short*)chunkRef->m_pBuffer, chunkRef->GetNumSamples()) != chunkRef->GetNumSamples())
 			{
@@ -103,7 +111,9 @@ int LibSndFileFile::ReadChunkMono(AudioChunkRef& chunk)
 		chunk.reset(new AudioChunk());
 		short temp[8000];
 		numRead = sf_read_short(m_pFile, temp, 8000);
-		chunk->SetBuffer(temp, sizeof(short)*numRead, AudioChunk::PcmAudio, 0, 0, m_sampleRate);
+		AudioChunkDetails details;
+		details.m_encoding = PcmAudio;
+		chunk->SetBuffer(temp, sizeof(short)*numRead, details);
 	}
 	else
 	{
diff --git a/orkaudio/audiofile/MediaChunkFile.cpp b/orkaudio/audiofile/MediaChunkFile.cpp
new file mode 100644
index 0000000..47b2e8b
--- /dev/null
+++ b/orkaudio/audiofile/MediaChunkFile.cpp
@@ -0,0 +1,154 @@
+/*
+ * Oreka -- A media capture and retrieval platform
+ * 
+ * Copyright (C) 2005, orecx LLC
+ *
+ * http://www.orecx.com
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License.
+ * Please refer to http://www.gnu.org/copyleft/gpl.html
+ *
+ */
+
+#include "MediaChunkFile.h"
+
+#define MAX_CHUNK_SIZE 100000
+
+
+MediaChunkFile::MediaChunkFile()
+{
+	m_stream = NULL;
+	m_mode = READ;
+	m_numChunksWritten = 0;
+	m_sampleRate = 0;
+}
+
+MediaChunkFile::~MediaChunkFile()
+{
+	Close();
+}
+
+
+void MediaChunkFile::Close()
+{
+	if(m_stream)
+	{
+		ACE_OS::fclose(m_stream);
+		m_stream = NULL;
+	}
+}
+
+void MediaChunkFile::WriteChunk(AudioChunkRef chunkRef)
+{
+	if(chunkRef.get() == NULL)
+	{
+		return;
+	}
+	if(chunkRef->GetDetails()->m_numBytes == 0)
+	{
+		return;
+	}
+
+	unsigned int numWritten = 0;
+	bool writeError = false;
+	if (m_stream)
+	{
+		int tmp = sizeof(AudioChunkDetails);
+		numWritten = ACE_OS::fwrite(chunkRef->GetDetails(), sizeof(AudioChunkDetails), 1, m_stream);
+		if(numWritten != 1)
+		{
+			writeError = true;
+		}
+		numWritten = ACE_OS::fwrite(chunkRef->m_pBuffer, sizeof(char), chunkRef->GetNumBytes(), m_stream);
+		if(numWritten != chunkRef->GetNumBytes())
+		{
+			writeError = true;
+		}
+	}
+	else
+	{
+		throw(CStdString("Write attempt on unopened file:")+ m_filename);
+	}
+
+	if (writeError)
+	{
+		throw(CStdString("Could not write to file:")+ m_filename);
+	}
+}
+
+int MediaChunkFile::ReadChunkMono(AudioChunkRef& chunkRef)
+{
+	unsigned int numRead = 0;
+	bool readError = false;
+
+	if (m_stream)
+	{
+		chunkRef.reset(new AudioChunk());
+		short temp[MAX_CHUNK_SIZE];
+		numRead = ACE_OS::fread(temp, sizeof(AudioChunkDetails), 1, m_stream);
+		if(numRead == 1)
+		{
+			AudioChunkDetails details;
+			memcpy(&details, temp, sizeof(AudioChunkDetails));
+
+			if(details.m_marker != MEDIA_CHUNK_MARKER)
+			{
+				throw(CStdString("Invalid marker in file:")+ m_filename);
+			}
+			if(details.m_numBytes >= MAX_CHUNK_SIZE)
+			{
+				throw(CStdString("Chunk too big in file:")+ m_filename);
+			}
+			else
+			{
+				numRead = ACE_OS::fread(temp, sizeof(char), details.m_numBytes, m_stream);
+				if(numRead != details.m_numBytes)
+				{
+					throw(CStdString("Incomplete chunk in file:")+ m_filename);
+				}
+				chunkRef->SetBuffer(temp, details.m_numBytes, details);
+			}
+		}
+	}
+	else
+	{
+		throw(CStdString("Read attempt on unopened file:")+ m_filename);
+	}
+	
+	return numRead;
+}
+
+
+void MediaChunkFile::Open(CStdString& filename, fileOpenModeEnum mode, bool stereo, int sampleRate)
+{
+	if(m_sampleRate == 0)
+	{
+		m_sampleRate = sampleRate;
+	}
+
+	if(!m_filename.Equals(filename))
+	{
+		m_filename = filename + GetExtension();
+	}
+	m_stream = NULL;
+	m_mode = mode;
+	if (mode == READ)
+	{
+		m_stream = ACE_OS::fopen((PCSTR)m_filename, "rb");
+	}
+	else
+	{
+		RecursiveMkdir(m_filename);
+		m_stream = ACE_OS::fopen((PCSTR)m_filename, "wb");
+	}
+	if(!m_stream)
+	{
+		throw(CStdString("Could not open file: ") + m_filename);
+	}
+}
+
+CStdString MediaChunkFile::GetExtension()
+{
+	return ".mcf";
+}
diff --git a/orkaudio/audiofile/MediaChunkFile.h b/orkaudio/audiofile/MediaChunkFile.h
new file mode 100644
index 0000000..b8e69ed
--- /dev/null
+++ b/orkaudio/audiofile/MediaChunkFile.h
@@ -0,0 +1,40 @@
+/*
+ * Oreka -- A media capture and retrieval platform
+ * 
+ * Copyright (C) 2005, orecx LLC
+ *
+ * http://www.orecx.com
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License.
+ * Please refer to http://www.gnu.org/copyleft/gpl.html
+ *
+ */
+
+#ifndef __MEDIACHUNKFILE_H__
+#define __MEDIACHUNKFILE_H__
+
+#include "audiofile/AudioFile.h"
+
+
+/** File class for saving audio chunks as-is */
+class MediaChunkFile : public AudioFile
+{
+public:
+	MediaChunkFile();
+	~MediaChunkFile();
+
+	void Open(CStdString& filename, fileOpenModeEnum mode, bool stereo = false, int sampleRate = 8000);
+	void Close();
+
+	void WriteChunk(AudioChunkRef chunkRef);
+	int ReadChunkMono(AudioChunkRef& chunkRef);
+
+	CStdString GetExtension();
+protected:
+
+	FILE* m_stream;
+};
+
+#endif
+
diff --git a/orkaudio/audiofile/PcmFile.cpp b/orkaudio/audiofile/PcmFile.cpp
index 0acfb3f..9d1a6d0 100644
--- a/orkaudio/audiofile/PcmFile.cpp
+++ b/orkaudio/audiofile/PcmFile.cpp
@@ -38,6 +38,15 @@ void PcmFile::Close()
 
 void PcmFile::WriteChunk(AudioChunkRef chunkRef)
 {
+	if(chunkRef.get() == NULL)
+	{
+		return;
+	}
+	if(chunkRef->GetDetails()->m_numBytes == 0)
+	{
+		return;
+	}
+
 	unsigned int numWritten = 0;
 	if (m_stream)
 	{
@@ -62,7 +71,9 @@ int PcmFile::ReadChunkMono(AudioChunkRef& chunkRef)
 		chunkRef.reset(new AudioChunk());
 		short temp[PCM_FILE_DEFAULT_CHUNK_NUM_SAMPLES];
 		numRead = ACE_OS::fread(temp, sizeof(short), PCM_FILE_DEFAULT_CHUNK_NUM_SAMPLES, m_stream);
-		chunkRef->SetBuffer(temp, sizeof(short)*numRead, AudioChunk::PcmAudio, 0, 0, m_sampleRate);
+		AudioChunkDetails details;
+		details.m_encoding = PcmAudio;
+		chunkRef->SetBuffer(temp, sizeof(short)*numRead, details);
 	}
 	else
 	{
diff --git a/orkaudio/filters/VoIpMixer/VoIpMixer.cpp b/orkaudio/filters/VoIpMixer/VoIpMixer.cpp
new file mode 100644
index 0000000..b2a7925
--- /dev/null
+++ b/orkaudio/filters/VoIpMixer/VoIpMixer.cpp
@@ -0,0 +1,353 @@
+/*
+ * Oreka -- A media capture and retrieval platform
+ * 
+ * Copyright (C) 2005, orecx LLC
+ *
+ * http://www.orecx.com
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License.
+ * Please refer to http://www.gnu.org/copyleft/gpl.html
+ *
+ */
+#pragma warning( disable: 4786 )
+
+#ifdef WIN32
+#define __CDECL__ __cdecl
+#else
+#define __CDECL__
+#endif
+#ifdef WIN32
+	#define DLL_EXPORT  __declspec( dllexport )
+#else
+	#define DLL_EXPORT
+#endif
+
+#include <queue>
+#include "Filter.h"
+#include "AudioCapture.h"
+extern "C"
+{
+#include "g711.h"
+}
+
+#define NUM_SAMPLES_CIRCULAR_BUFFER 8000
+#define NUM_SAMPLES_TRIGGER 4000			// when we have this number of available samples make a shipment
+#define NUM_SAMPLES_SHIPMENT_HOLDOFF 2000	// when shipping, ship everything but this number of samples 
+
+
+class VoIpMixer : public Filter
+{
+public:
+	VoIpMixer();
+
+	FilterRef __CDECL__ Instanciate();
+	void __CDECL__ AudioChunkIn(AudioChunkRef& chunk);
+	void __CDECL__ AudioChunkOut(AudioChunkRef& chunk);
+	AudioEncodingEnum __CDECL__ GetInputAudioEncoding();
+	AudioEncodingEnum __CDECL__ GetOutputAudioEncoding();
+	CStdString __CDECL__ GetName();
+	int __CDECL__ GetInputRtpPayloadType();
+
+private:
+	//AudioChunkRef m_outputAudioChunk;
+	std::queue<AudioChunkRef> m_outputQueue;
+
+	void StoreRtpPacket(AudioChunkRef& chunk);
+	void CreateShipment(size_t silenceSize);
+	unsigned int FreeSpace();
+	unsigned int UsedSpace();
+	short* GetHoldOffPtr();
+	short* CircularPointerAddOffset(short *ptr, size_t offset);
+	short* CicularPointerSubtractOffset(short *ptr, size_t offset);
+
+	short* m_writePtr;		// pointer after newest RTP data we've received
+	short* m_readPtr;		// where to read from next
+	unsigned int m_readTimestamp;	// timestamp that the next shipment will have
+	unsigned int m_writeTimestamp;	// timestamp that the next RTP buffer should have
+	short* m_bufferEnd;
+	short m_buffer[NUM_SAMPLES_CIRCULAR_BUFFER];
+	unsigned int m_shippedSamples;
+
+};
+
+VoIpMixer::VoIpMixer()
+{
+	m_writePtr = m_buffer;
+	m_readPtr = m_buffer;
+	m_bufferEnd = m_buffer + NUM_SAMPLES_CIRCULAR_BUFFER;
+	m_writeTimestamp = 0;
+	m_readTimestamp = 0;
+	//m_log = Logger::getLogger("rtpringbuffer");
+	m_shippedSamples = 0;
+}
+
+FilterRef VoIpMixer::Instanciate()
+{
+	FilterRef Filter(new VoIpMixer());
+	return Filter;
+}
+
+void VoIpMixer::AudioChunkIn(AudioChunkRef& chunk)
+{	
+	AudioChunkDetails* details = chunk->GetDetails();		
+	if(details->m_encoding != PcmAudio)
+	{
+		throw (CStdString("VoIpMixer input audio must be PCM !"));
+	}
+
+	unsigned int rtpEndTimestamp = details->m_timestamp + chunk->GetNumSamples();  // GetNumSamples()  #############################
+
+	if(m_writeTimestamp == 0)
+	{
+		// First RTP packet of the session
+		//LOG4CXX_DEBUG(m_log, m_capturePort + " first packet");
+		m_writeTimestamp = details->m_timestamp;
+		m_readTimestamp = m_writeTimestamp;
+		StoreRtpPacket(chunk);
+	}
+	else if (details->m_timestamp >= m_readTimestamp)	// drop packets that are older than last shipment
+	{
+		if( (int)(rtpEndTimestamp - m_writeTimestamp) <= (int)FreeSpace())
+		{
+			// RTP packet fits into current buffer
+			StoreRtpPacket(chunk);
+
+			if(UsedSpace() > NUM_SAMPLES_TRIGGER)
+			{
+				// We have enough stuff, make a shipment
+				CreateShipment(0);
+			}
+		}
+		else
+		{
+			// RTP packet does not fit into current buffer
+			// work out how much silence we need to add to the current buffer when shipping
+			size_t silenceSize = details->m_timestamp - m_writeTimestamp;
+			CreateShipment(silenceSize);
+
+			// reset buffer
+			m_writePtr = m_buffer;
+			m_readPtr = m_buffer;
+			m_writeTimestamp = details->m_timestamp;
+			m_readTimestamp = m_writeTimestamp;
+
+			// Store new packet
+			StoreRtpPacket(chunk);
+		}
+	}
+	else
+	{
+		//LOG4CXX_DEBUG(m_log, m_capturePort + " packet too old, dropped");
+	}
+	CStdString debug;
+	debug.Format("free:%u used:%u wr:%x rd:%x wrts:%u rdts:%d", FreeSpace(), UsedSpace(), m_writePtr, m_readPtr, m_writeTimestamp, m_readTimestamp);
+	//LOG4CXX_DEBUG(m_log, debug);
+}
+
+void VoIpMixer::AudioChunkOut(AudioChunkRef& chunk)
+{
+	if(m_outputQueue.size() > 0)
+	{
+		chunk = m_outputQueue.front();
+		m_outputQueue.pop();
+	}
+	else
+	{
+		chunk.reset();
+	}
+}
+
+AudioEncodingEnum VoIpMixer::GetInputAudioEncoding()
+{
+	return PcmAudio;
+}
+
+AudioEncodingEnum VoIpMixer::GetOutputAudioEncoding()
+{
+	return PcmAudio;
+}
+
+CStdString VoIpMixer::GetName()
+{
+	return "VoIpMixer";
+}
+
+
+int VoIpMixer::GetInputRtpPayloadType(void)	// does not link if not defined here ?
+{
+	return -1;
+}
+
+// Writes to the internal buffer without any size verification
+void VoIpMixer::StoreRtpPacket(AudioChunkRef& audioChunk)
+{
+	CStdString debug;
+	AudioChunkDetails* details = audioChunk->GetDetails();
+
+	// 1. Silence from write pointer until end of RTP packet
+	unsigned int endRtpTimestamp = details->m_timestamp + audioChunk->GetNumSamples();	  // GetNumSamples()  #############################
+	if (endRtpTimestamp > m_writeTimestamp)
+	{
+		for(int i=0; i<(endRtpTimestamp - m_writeTimestamp); i++)
+		{
+			*m_writePtr = 0;
+			m_writePtr++;
+			if(m_writePtr >= m_bufferEnd)
+			{
+				m_writePtr = m_buffer;
+			}
+		}
+		int silenceSize = endRtpTimestamp - m_writeTimestamp;
+		m_writeTimestamp = endRtpTimestamp;
+		debug.Format("Zeroed %d samples, wr:%x wrts:%u", silenceSize, m_writePtr, m_writeTimestamp);
+		//LOG4CXX_DEBUG(m_log, debug);
+	}
+
+	// 2. Mix in the latest samples from this RTP packet
+	unsigned int timestampDelta = m_writeTimestamp - details->m_timestamp;
+	ASSERT(timestampDelta>=0);
+	short* tempWritePtr = CicularPointerSubtractOffset(m_writePtr, timestampDelta);
+	short* payload = (short *)audioChunk->m_pBuffer;	// payload should be short* #################################################
+
+	for(int i=0; i<audioChunk->GetNumSamples() ; i++)	//#################################################
+	{
+		*tempWritePtr += payload[i];
+		tempWritePtr++;
+		if(tempWritePtr >= m_bufferEnd)
+		{
+			tempWritePtr = m_buffer;
+		}
+	}
+	debug.Format("Copied %d samples, tmpwr:%x", audioChunk->GetNumSamples(), tempWritePtr);	//#################################################
+	//LOG4CXX_DEBUG(m_log, debug);
+}
+
+short* VoIpMixer::CircularPointerAddOffset(short *ptr, size_t offset)
+{
+	if((ptr + offset) >= m_bufferEnd)
+	{
+		return m_buffer + offset - (m_bufferEnd-ptr);
+	}
+	else
+	{
+		return ptr + offset;
+	}
+}
+
+short* VoIpMixer::CicularPointerSubtractOffset(short *ptr, size_t offset)
+{
+	if((ptr-offset) < m_buffer)
+	{
+		return m_bufferEnd - offset + (ptr-m_buffer);
+	}
+	else
+	{
+		return ptr - offset;
+	}
+}
+
+void VoIpMixer::CreateShipment(size_t silenceSize)
+{
+	// 1. ship from readPtr until stop pointer or until end of buffer if wrapped
+	bool bufferWrapped = false;
+	short* stopPtr = NULL;
+	short* wrappedStopPtr = NULL;
+	if (silenceSize)
+	{
+		// There is additional silence to ship, do not take holdoff into account
+		stopPtr = m_writePtr;
+	}
+	else
+	{
+		stopPtr = CicularPointerSubtractOffset(m_writePtr, NUM_SAMPLES_SHIPMENT_HOLDOFF);
+	}
+
+	if (stopPtr < m_readPtr)
+	{
+		wrappedStopPtr = stopPtr;
+		stopPtr = m_bufferEnd;
+		bufferWrapped = true;
+	}
+	size_t shortSize = stopPtr-m_readPtr;
+	size_t byteSize = shortSize*2;
+	AudioChunkRef chunk(new AudioChunk());
+	AudioChunkDetails details;
+	details.m_encoding = PcmAudio;
+	chunk->SetBuffer((void*)m_readPtr, byteSize, details);
+	m_outputQueue.push(chunk);
+	m_shippedSamples += shortSize;
+	m_readPtr = CircularPointerAddOffset(m_readPtr ,shortSize);
+	m_readTimestamp += shortSize;
+
+	CStdString debug;
+	debug.Format("Ship %d samples, rd:%x rdts:%u", shortSize, m_readPtr, m_readTimestamp);
+	//LOG4CXX_DEBUG(m_log, debug);
+
+
+	// 2. ship from beginning of buffer until stop ptr
+	if(bufferWrapped) 
+	{
+		shortSize = wrappedStopPtr - m_buffer;
+		byteSize = shortSize*2;
+		chunk.reset(new AudioChunk());
+		AudioChunkDetails details;
+		details.m_encoding = PcmAudio;
+		chunk->SetBuffer((void*)m_buffer, byteSize, details);
+		m_outputQueue.push(chunk);
+		m_shippedSamples += shortSize;
+		m_readPtr = CircularPointerAddOffset(m_readPtr ,shortSize);
+		m_readTimestamp += shortSize;
+		debug.Format("Ship wrapped %d samples, rd:%x rdts:%u", shortSize, m_readPtr, m_readTimestamp);
+		//LOG4CXX_DEBUG(m_log, debug);
+	}
+
+	// 3. ship silence
+	if (silenceSize)
+	{
+		byteSize = silenceSize*2;
+		char* silenceBuffer = (char*)calloc(byteSize, 1);
+		if (silenceBuffer)
+		{
+			AudioChunkRef chunk(new AudioChunk());
+			AudioChunkDetails details;
+			details.m_encoding = PcmAudio;
+			chunk->SetBuffer((void*)silenceBuffer, byteSize, details);
+			m_outputQueue.push(chunk);
+			m_shippedSamples += silenceSize;
+			m_readPtr = CircularPointerAddOffset(m_readPtr ,silenceSize);
+			m_readTimestamp += silenceSize;
+		}
+		debug.Format("Ship %d silence samples, rd:%x rdts:%u", silenceSize, m_readPtr, m_readTimestamp);
+		//LOG4CXX_DEBUG(m_log, debug);
+	}
+}
+
+
+unsigned int VoIpMixer::UsedSpace()
+{
+	if(m_writePtr >= m_readPtr)
+	{
+		return m_writePtr - m_readPtr;
+	}
+	return NUM_SAMPLES_CIRCULAR_BUFFER + m_writePtr - m_readPtr;
+}
+
+
+unsigned int VoIpMixer::FreeSpace()
+{
+	return NUM_SAMPLES_CIRCULAR_BUFFER - UsedSpace();
+}
+
+
+//=====================================================================
+
+
+extern "C"
+{
+	DLL_EXPORT void __CDECL__ Initialize()
+	{
+		FilterRef filter(new VoIpMixer());
+		FilterRegistry::instance()->RegisterFilter(filter);
+	}
+}
\ No newline at end of file
diff --git a/orkaudio/filters/VoIpMixer/VoIpMixer.dsp b/orkaudio/filters/VoIpMixer/VoIpMixer.dsp
new file mode 100644
index 0000000..2d43dd8
--- /dev/null
+++ b/orkaudio/filters/VoIpMixer/VoIpMixer.dsp
@@ -0,0 +1,162 @@
+# Microsoft Developer Studio Project File - Name="VoIpMixer" - Package Owner=<4>
+# Microsoft Developer Studio Generated Build File, Format Version 6.00
+# ** DO NOT EDIT **
+
+# TARGTYPE "Win32 (x86) Dynamic-Link Library" 0x0102
+
+CFG=VoIpMixer - Win32 Debug
+!MESSAGE This is not a valid makefile. To build this project using NMAKE,
+!MESSAGE use the Export Makefile command and run
+!MESSAGE 
+!MESSAGE NMAKE /f "VoIpMixer.mak".
+!MESSAGE 
+!MESSAGE You can specify a configuration when running NMAKE
+!MESSAGE by defining the macro CFG on the command line. For example:
+!MESSAGE 
+!MESSAGE NMAKE /f "VoIpMixer.mak" CFG="VoIpMixer - Win32 Debug"
+!MESSAGE 
+!MESSAGE Possible choices for configuration are:
+!MESSAGE 
+!MESSAGE "VoIpMixer - Win32 Release" (based on "Win32 (x86) Dynamic-Link Library")
+!MESSAGE "VoIpMixer - Win32 Debug" (based on "Win32 (x86) Dynamic-Link Library")
+!MESSAGE 
+
+# Begin Project
+# PROP AllowPerConfigDependencies 0
+# PROP Scc_ProjName ""
+# PROP Scc_LocalPath ""
+CPP=cl.exe
+MTL=midl.exe
+RSC=rc.exe
+
+!IF  "$(CFG)" == "VoIpMixer - Win32 Release"
+
+# PROP BASE Use_MFC 0
+# PROP BASE Use_Debug_Libraries 0
+# PROP BASE Output_Dir "Release"
+# PROP BASE Intermediate_Dir "Release"
+# PROP BASE Target_Dir ""
+# PROP Use_MFC 0
+# PROP Use_Debug_Libraries 0
+# PROP Output_Dir "Release"
+# PROP Intermediate_Dir "Release"
+# PROP Target_Dir ""
+# ADD BASE CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D "VoIpMixer_EXPORTS" /YX /FD /c
+# ADD CPP /nologo /MT /W3 /GX /O2 /I "C:\devExt\winpcap\WpdPack\Include" /I "../.." /I "../../../OrkBaseCxx" /I "C:\devExt\boost\boost_1_32_0" /I "C:\devExt\ACE_wrappers" /I "C:\devExt\xerces++\xerces-c_2_6_0-windows_nt-msvc_60\include" /I "../Common" /I "C:\devExt\log4cxx\log4cxx-0.9.7\include" /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D "VoIpMixer_EXPORTS" /YX /FD /c
+# ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /win32
+# ADD MTL /nologo /D "NDEBUG" /mktyplib203 /win32
+# ADD BASE RSC /l 0x809 /d "NDEBUG"
+# ADD RSC /l 0x809 /d "NDEBUG"
+BSC32=bscmake.exe
+# ADD BASE BSC32 /nologo
+# ADD BSC32 /nologo
+LINK32=link.exe
+# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /machine:I386
+# ADD LINK32 packet.lib wpcap.lib wininet.lib OrkBase.lib ace.lib log4cxx.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /machine:I386 /libpath:"C:\devExt\winpcap\WpdPack\Lib" /libpath:"../../../OrkBaseCxx/Release" /libpath:"C:\devExt\log4cxx\log4cxx-0.9.7\msvc\Lib\Release" /libpath:"C:\devExt\ACE_wrappers\lib"
+
+!ELSEIF  "$(CFG)" == "VoIpMixer - Win32 Debug"
+
+# PROP BASE Use_MFC 0
+# PROP BASE Use_Debug_Libraries 1
+# PROP BASE Output_Dir "Debug"
+# PROP BASE Intermediate_Dir "Debug"
+# PROP BASE Target_Dir ""
+# PROP Use_MFC 0
+# PROP Use_Debug_Libraries 1
+# PROP Output_Dir "Debug"
+# PROP Intermediate_Dir "Debug"
+# PROP Ignore_Export_Lib 0
+# PROP Target_Dir ""
+# ADD BASE CPP /nologo /MTd /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D "VoIpMixer_EXPORTS" /YX /FD /GZ /c
+# ADD CPP /nologo /MDd /W3 /Gm /GX /ZI /Od /I "C:\devExt\winpcap\WpdPack\Include" /I "../.." /I "../../../OrkBaseCxx" /I "C:\devExt\boost\boost_1_32_0" /I "C:\devExt\ACE_wrappers" /I "C:\devExt\xerces++\xerces-c_2_6_0-windows_nt-msvc_60\include" /I "../Common" /I "C:\devExt\log4cxx\log4cxx-0.9.7\include" /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D "VoIpMixer_EXPORTS" /YX /FD /GZ /c
+# ADD BASE MTL /nologo /D "_DEBUG" /mktyplib203 /win32
+# ADD MTL /nologo /D "_DEBUG" /mktyplib203 /win32
+# ADD BASE RSC /l 0x809 /d "_DEBUG"
+# ADD RSC /l 0x809 /d "_DEBUG"
+BSC32=bscmake.exe
+# ADD BASE BSC32 /nologo
+# ADD BSC32 /nologo
+LINK32=link.exe
+# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /debug /machine:I386 /pdbtype:sept
+# ADD LINK32 packet.lib wpcap.lib wininet.lib OrkBaseD.lib aced.lib log4cxx.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /debug /machine:I386 /pdbtype:sept /libpath:"C:\devExt\winpcap\WpdPack\Lib" /libpath:"../../../OrkBaseCxx/Debug" /libpath:"C:\devExt\log4cxx\log4cxx-0.9.7\msvc\Lib\Debug" /libpath:"C:\devExt\ACE_wrappers\lib"
+
+!ENDIF 
+
+# Begin Target
+
+# Name "VoIpMixer - Win32 Release"
+# Name "VoIpMixer - Win32 Debug"
+# Begin Group "Source Files"
+
+# PROP Default_Filter "cpp;c;cxx;rc;def;r;odl;idl;hpj;bat"
+# Begin Source File
+
+SOURCE=..\..\AudioCapturePlugin.h
+# End Source File
+# Begin Source File
+
+SOURCE=..\Common\AudioCapturePluginCommon.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=..\Common\AudioCapturePluginCommon.h
+# End Source File
+# Begin Source File
+
+SOURCE=..\..\ConfigManager.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\g711.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\g711.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\PacketHeaderDefs.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=.\PacketHeaderDefs.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\Rtp.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=.\Rtp.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\RtpSession.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=.\RtpSession.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\VoIpMixer.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=.\VoIpMixerConfig.cpp
+# End Source File
+# Begin Source File
+
+SOURCE=.\VoIpMixerConfig.h
+# End Source File
+# End Group
+# Begin Group "Header Files"
+
+# PROP Default_Filter "h;hpp;hxx;hm;inl"
+# End Group
+# Begin Group "Resource Files"
+
+# PROP Default_Filter "ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe"
+# End Group
+# End Target
+# End Project