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

2 files changed, 27 insertions, 4 deletions

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;
 	}
 }