From 46f80e2a15a8f0d6cbe9b4b5a19d4a6650ad95d5 Mon Sep 17 00:00:00 2001
From: f4exb <f4exb06@gmail.com>
Date: Fri, 9 Mar 2018 06:54:45 +0100
Subject: [PATCH] DSD demod: implement RTP over UDP for audio copy final

---
 plugins/channelrx/demoddsd/dsddemodgui.cpp |  4 ++++
 sdrbase/audio/audiofifo.cpp                | 17 +++++++++++------
 2 files changed, 15 insertions(+), 6 deletions(-)

diff --git a/plugins/channelrx/demoddsd/dsddemodgui.cpp b/plugins/channelrx/demoddsd/dsddemodgui.cpp
index 066b470e1..ec6966bd4 100644
--- a/plugins/channelrx/demoddsd/dsddemodgui.cpp
+++ b/plugins/channelrx/demoddsd/dsddemodgui.cpp
@@ -308,6 +308,10 @@ DSDDemodGUI::DSDDemodGUI(PluginAPI* pluginAPI, DeviceUISet *deviceUISet, Baseban
 	m_deviceUISet->addChannelMarker(&m_channelMarker);
 	m_deviceUISet->addRollupWidget(this);
 
+    if (!m_dsdDemod->isAudioNetSinkRTPCapable()) {
+        ui->useRTP->hide();
+    }
+
 	connect(&m_channelMarker, SIGNAL(changedByCursor()), this, SLOT(channelMarkerChangedByCursor()));
     connect(&m_channelMarker, SIGNAL(highlightedByCursor()), this, SLOT(channelMarkerHighlightedByCursor()));
 
diff --git a/sdrbase/audio/audiofifo.cpp b/sdrbase/audio/audiofifo.cpp
index 1a412d454..efd24a9e6 100644
--- a/sdrbase/audio/audiofifo.cpp
+++ b/sdrbase/audio/audiofifo.cpp
@@ -78,12 +78,6 @@ uint AudioFifo::write(const quint8* data, uint32_t numSamples, int timeout_ms)
 	uint32_t remaining;
 	uint32_t copyLen;
 
-	if (m_copyToUDP && m_audioNetSink)
-	{
-	    //m_udpSink->write((AudioSample *) data, numSamples);
-	    m_audioNetSink->write((AudioSample *) data, numSamples);
-	}
-
 	if(m_fifo == 0)
 	{
 		return 0;
@@ -223,6 +217,17 @@ uint AudioFifo::read(quint8* data, uint32_t numSamples, int timeout_ms)
 		copyLen = MIN(remaining, m_fill);
 		copyLen = MIN(copyLen, m_size - m_head);
 		memcpy(data, m_fifo + (m_head * m_sampleSize), copyLen * m_sampleSize);
+
+	    if (m_copyToUDP && m_audioNetSink)
+	    {
+	        for (quint8 *p = data; p < data + copyLen* m_sampleSize;)
+	        {
+	            AudioSample *a = (AudioSample *) p;
+	            m_audioNetSink->write((a->l + a->r)/2);
+	            p += m_sampleSize;
+	        }
+	    }
+
 		m_head += copyLen;
 		m_head %= m_size;
 		m_fill -= copyLen;