RTP audio sink fixes and test in NFM demod (endianess problem)

plugins/channelrx/demodnfm/nfmdemod.cpp

@@ -83,6 +83,12 @@ NFMDemod::NFMDemod(DeviceSourceAPI *devieAPI) :
 	m_audioNetSink = new AudioNetSink(this);
 	m_audioNetSink->setDestination(m_settings.m_udpAddress, m_settings.m_udpPort);
 
+	if (m_audioNetSink->selectType(AudioNetSink::SinkRTP)) {
+	    qDebug("NFMDemod::NFMDemod: set audio sink to RTP mode");
+	} else {
+	    qWarning("NFMDemod::NFMDemod: RTP support for audio sink not available. Fall back too UDP");
+	}
+
     m_channelizer = new DownChannelizer(this);
     m_threadedChannelizer = new ThreadedBasebandSampleSink(m_channelizer, this);
     m_deviceAPI->addThreadedSink(m_threadedChannelizer);

plugins/channelrx/demodnfm/nfmdemodsettings.cpp

@@ -52,7 +52,7 @@ void NFMDemodSettings::resetToDefaults()
     m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
     m_copyAudioToUDP = false;
     m_udpAddress = "127.0.0.1";
-    m_udpPort = 9999;
+    m_udpPort = 9998;
     m_rgbColor = QColor(255, 0, 0).rgb();
     m_title = "NFM Demodulator";
 }

sdrbase/CMakeLists.txt

@@ -216,6 +216,10 @@ if (JRTPLIB_FOUND)
         ${sdrbase_HEADERS}
         util/rtpsink.h
     )
+    set(sdrbase_SOURCES
+        ${sdrbase_SOURCES}
+        util/rtpsink.cpp
+    )
     add_definitions(-DHAS_JRTPLIB)
     include_directories(${JRTPLIB_INCLUDE_DIR})
 endif(JRTPLIB_FOUND)

sdrbase/audio/audionetsink.cpp

@@ -26,33 +26,33 @@ const int AudioNetSink::m_udpBlockSize = 512;
 AudioNetSink::AudioNetSink(QObject *parent, bool stereo) :
     m_type(SinkUDP),
     m_udpBufferAudioMono(0),
-    m_udpBufferAudioStereo(0)
+    m_udpBufferAudioStereo(0),
+    m_rtpBufferAudio(0)
 {
-    if (stereo)
+    if (stereo) {
-    {
         m_udpBufferAudioStereo = new UDPSink<AudioSample>(parent, m_udpBlockSize);
-#ifdef HAS_JRTPLIB
+    } else {
-        m_rtpBufferAudioStereo = new RTPSink<AudioSample>("127.0.0.1", 9999, 48000);
-        m_rtpBufferAudioMono = 0;
-#endif
-    }
-    else
-    {
         m_udpBufferAudioMono = new UDPSink<int16_t>(parent, m_udpBlockSize);
-#ifdef HAS_JRTPLIB
-        m_rtpBufferAudioMono = new RTPSink<int16_t>("127.0.0.1", 9999, 48000);
-        m_rtpBufferAudioStereo = 0;
-#endif
     }
+
+#ifdef HAS_JRTPLIB
+    m_rtpBufferAudio = new RTPSink("127.0.0.1", 9999, stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono);
+#endif
 }
 
 AudioNetSink::~AudioNetSink()
 {
-    if (m_udpBufferAudioMono) { delete m_udpBufferAudioMono; }
+    if (m_udpBufferAudioMono) {
-    if (m_udpBufferAudioStereo) { delete m_udpBufferAudioStereo; }
+        delete m_udpBufferAudioMono;
+    }
+
+    if (m_udpBufferAudioStereo) {
+        delete m_udpBufferAudioStereo;
+    }
 #ifdef HAS_JRTPLIB
-    if (m_rtpBufferAudioMono) { delete m_rtpBufferAudioMono; }
+    if (m_rtpBufferAudio) {
-    if (m_rtpBufferAudioStereo) { delete m_rtpBufferAudioStereo; }
+        delete m_rtpBufferAudio;
+    }
 #endif
 }
 
@@ -87,51 +87,39 @@ void AudioNetSink::setDestination(const QString& address, uint16_t port)
     if (m_udpBufferAudioStereo) {
         m_udpBufferAudioStereo->setDestination(address, port);
     }
+
 #ifdef HAS_JRTPLIB
-    if (m_rtpBufferAudioMono) {
+    if (m_rtpBufferAudio) {
-        m_rtpBufferAudioMono->setDestination(address, port);
+        m_rtpBufferAudio->setDestination(address, port);
-    }
-    if (m_rtpBufferAudioStereo) {
-        m_rtpBufferAudioStereo->setDestination(address, port);
     }
 #endif
 }
 
+#ifdef HAS_JRTPLIB
 void AudioNetSink::addDestination(const QString& address, uint16_t port)
 {
-#ifdef HAS_JRTPLIB
+    if (m_rtpBufferAudio) {
-    if (m_rtpBufferAudioMono) {
+        m_rtpBufferAudio->addDestination(address, port);
-        m_rtpBufferAudioMono->addDestination(address, port);
     }
-    if (m_rtpBufferAudioStereo) {
-        m_rtpBufferAudioStereo->addDestination(address, port);
-    }
-#endif
 }
+#else
+void AudioNetSink::addDestination(const QString& address __attribute__((unused)), uint16_t port __attribute__((unused)))
+{
+}
+#endif
 
+#ifdef HAS_JRTPLIB
 void AudioNetSink::deleteDestination(const QString& address, uint16_t port)
 {
-#ifdef HAS_JRTPLIB
+    if (m_rtpBufferAudio) {
-    if (m_rtpBufferAudioMono) {
+        m_rtpBufferAudio->deleteDestination(address, port);
-        m_rtpBufferAudioMono->deleteDestination(address, port);
     }
-    if (m_rtpBufferAudioStereo) {
-        m_rtpBufferAudioStereo->deleteDestination(address, port);
-    }
-#endif
 }
-
+#else
-void AudioNetSink::setSampleRate(int sampleRate)
+void AudioNetSink::deleteDestination(const QString& address __attribute__((unused)), uint16_t port __attribute__((unused)))
 {
-#ifdef HAS_JRTPLIB
-    if (m_rtpBufferAudioMono) {
-        m_rtpBufferAudioMono->setSampleRate(sampleRate);
-    }
-    if (m_rtpBufferAudioStereo) {
-        m_rtpBufferAudioStereo->setSampleRate(sampleRate);
-    }
-#endif
 }
+#endif
 
 void AudioNetSink::write(qint16 sample)
 {
@@ -143,6 +131,7 @@ void AudioNetSink::write(qint16 sample)
         m_udpBufferAudioMono->write(sample);
     } else if (m_type == SinkRTP) {
 #ifdef HAS_JRTPLIB
+        m_rtpBufferAudio->write((uint8_t *) &sample);
 #endif
     }
 }
@@ -157,6 +146,7 @@ void AudioNetSink::write(const AudioSample& sample)
         m_udpBufferAudioStereo->write(sample);
     } else if (m_type == SinkRTP) {
 #ifdef HAS_JRTPLIB
+        m_rtpBufferAudio->write((uint8_t *) &sample);
 #endif
     }
 }

sdrbase/audio/audionetsink.h

@@ -23,7 +23,7 @@
 #include "util/export.h"
 
 template<typename T> class UDPSink;
-template<typename T> class RTPSink;
+class RTPSink;
 
 class SDRANGEL_API AudioNetSink {
 public:
@@ -39,7 +39,6 @@ public:
     void setDestination(const QString& address, uint16_t port);
     void addDestination(const QString& address, uint16_t port);
     void deleteDestination(const QString& address, uint16_t port);
-    void setSampleRate(int sampleRate);
 
     void write(qint16 sample);
     void write(const AudioSample& sample);
@@ -52,8 +51,7 @@ protected:
     SinkType m_type;
     UDPSink<qint16> *m_udpBufferAudioMono;
     UDPSink<AudioSample> *m_udpBufferAudioStereo;
-    RTPSink<qint16> *m_rtpBufferAudioMono;
+    RTPSink *m_rtpBufferAudio;
-    RTPSink<AudioSample> *m_rtpBufferAudioStereo;
 };
 
 

sdrbase/util/rtpsink.cpp

@@ -0,0 +1,220 @@
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2018 F4EXB                                                      //
+// written by Edouard Griffiths                                                  //
+//                                                                               //
+// This program is free software; you can redistribute it and/or modify          //
+// it under the terms of the GNU General Public License as published by          //
+// the Free Software Foundation as version 3 of the License, or                  //
+//                                                                               //
+// This program is distributed in the hope that it will be useful,               //
+// but WITHOUT ANY WARRANTY; without even the implied warranty of                //
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the                  //
+// GNU General Public License V3 for more details.                               //
+//                                                                               //
+// You should have received a copy of the GNU General Public License             //
+// along with this program. If not, see <http://www.gnu.org/licenses/>.          //
+///////////////////////////////////////////////////////////////////////////////////
+
+#include "rtpsink.h"
+#include "dsp/dsptypes.h"
+
+RTPSink::RTPSink(const QString& address, uint16_t port, PayloadType payloadType) :
+    m_sampleRate(48000),
+    m_sampleBytes(0),
+    m_packetSamples(0),
+    m_bufferSize(0),
+    m_sampleBufferIndex(0),
+    m_byteBuffer(0),
+    m_destport(port),
+    m_mutex(QMutex::Recursive)
+{
+    qDebug("RTPSink::RTPSink");
+    m_destip = inet_addr(address.toStdString().c_str());
+    m_destip = ntohl(m_destip);
+
+    m_rtpSessionParams.SetOwnTimestampUnit(1.0 / (double) m_sampleRate);
+    int status = m_rtpSession.Create(m_rtpSessionParams);
+
+    if (status < 0) {
+        qCritical("RTPSink::RTPSink: cannot create session: %s", jrtplib::RTPGetErrorString(status).c_str());
+    } else {
+        qDebug("RTPSink::RTPSink: created session: %s", jrtplib::RTPGetErrorString(status).c_str());
+    }
+
+    setPayloadType(payloadType);
+}
+
+RTPSink::~RTPSink()
+{
+    jrtplib::RTPTime delay = jrtplib::RTPTime(10.0);
+    m_rtpSession.BYEDestroy(delay, "Time's up", 9);
+
+    if (m_byteBuffer) {
+        delete[] m_byteBuffer;
+    }
+}
+
+void RTPSink::setPayloadType(PayloadType payloadType)
+{
+    QMutexLocker locker(&m_mutex);
+
+    qDebug("RTPSink::setPayloadType: %d", payloadType);
+
+    switch (payloadType)
+    {
+    case PayloadL16Stereo:
+        m_sampleRate = 48000;
+        m_sampleBytes = sizeof(AudioSample);
+        m_rtpSession.SetDefaultPayloadType(96);
+        break;
+    case PayloadL16Mono:
+    default:
+        m_sampleRate = 48000;
+        m_sampleBytes = sizeof(int16_t);
+        m_rtpSession.SetDefaultPayloadType(96);
+        break;
+    }
+
+    m_packetSamples = m_sampleRate/50; // 20ms packet samples
+    m_bufferSize = m_packetSamples * m_sampleBytes;
+
+    if (m_byteBuffer) {
+        delete[] m_byteBuffer;
+    }
+
+    m_byteBuffer = new uint8_t[m_bufferSize];
+    m_sampleBufferIndex = 0;
+    m_payloadType = payloadType;
+
+    int status = m_rtpSession.SetTimestampUnit(1.0 / (double) m_sampleRate);
+
+    if (status < 0) {
+        qCritical("RTPSink::setPayloadType: cannot set timestamp unit: %s", jrtplib::RTPGetErrorString(status).c_str());
+    } else {
+        qDebug("RTPSink::setPayloadType: timestamp unit set to %f: %s",
+               1.0 / (double) m_sampleRate,
+               jrtplib::RTPGetErrorString(status).c_str());
+    }
+
+    status = m_rtpSession.SetDefaultMark(false);
+
+    if (status < 0) {
+        qCritical("RTPSink::setPayloadType: cannot set default mark: %s", jrtplib::RTPGetErrorString(status).c_str());
+    } else {
+        qDebug("RTPSink::setPayloadType: set default mark to false: %s", jrtplib::RTPGetErrorString(status).c_str());
+    }
+
+    status = m_rtpSession.SetDefaultTimestampIncrement(m_packetSamples);
+
+    if (status < 0) {
+        qCritical("RTPSink::setPayloadType: cannot set default timestamp increment: %s", jrtplib::RTPGetErrorString(status).c_str());
+    } else {
+        qDebug("RTPSink::setPayloadType: set default timestamp increment to %d: %s", m_packetSamples, jrtplib::RTPGetErrorString(status).c_str());
+    }
+
+    status = m_rtpSession.SetMaximumPacketSize(m_bufferSize+40);
+
+    if (status < 0) {
+        qCritical("RTPSink::setPayloadType: cannot set maximum packet size: %s", jrtplib::RTPGetErrorString(status).c_str());
+    } else {
+        qDebug("RTPSink::setPayloadType: set maximum packet size to %d bytes: %s", m_bufferSize+40, jrtplib::RTPGetErrorString(status).c_str());
+    }
+}
+
+void RTPSink::setDestination(const QString& address, uint16_t port)
+{
+    m_rtpSession.ClearDestinations();
+    m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
+    m_destip = inet_addr(address.toStdString().c_str());
+    m_destip = ntohl(m_destip);
+    m_destport = port;
+
+    int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
+
+    if (status < 0) {
+        qCritical("RTPSink::setDestination: cannot set destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
+    }
+}
+
+void RTPSink::deleteDestination(const QString& address, uint16_t port)
+{
+    uint32_t destip = inet_addr(address.toStdString().c_str());
+    destip = ntohl(m_destip);
+
+    int status =  m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(destip, port));
+
+    if (status < 0) {
+        qCritical("RTPSink::deleteDestination: cannot delete destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
+    }
+}
+
+void RTPSink::addDestination(const QString& address, uint16_t port)
+{
+    uint32_t destip = inet_addr(address.toStdString().c_str());
+    destip = ntohl(m_destip);
+
+    int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(destip, port));
+
+    if (status < 0) {
+        qCritical("RTPSink::addDestination: cannot add destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
+    } else {
+        qDebug("RTPSink::addDestination: destination address set to %s:%d: %s",
+                address.toStdString().c_str(),
+                port,
+                jrtplib::RTPGetErrorString(status).c_str());
+    }
+}
+
+void RTPSink::write(uint8_t *sampleByte)
+{
+    QMutexLocker locker(&m_mutex);
+
+    if (m_sampleBufferIndex < m_packetSamples)
+    {
+        memcpy(&m_byteBuffer[m_sampleBufferIndex*m_sampleBytes], sampleByte, m_sampleBytes);
+        m_sampleBufferIndex++;
+    }
+    else
+    {
+        int status = m_rtpSession.SendPacket((const void *) m_byteBuffer, (std::size_t) m_bufferSize);
+
+        if (status < 0) {
+            qCritical("RTPSink::write: cannot write packet: %s", jrtplib::RTPGetErrorString(status).c_str());
+        }
+
+        memcpy(&m_byteBuffer[0], sampleByte, m_sampleBytes);
+        m_sampleBufferIndex = 1;
+    }
+}
+
+void RTPSink::write(uint8_t *samples, int nbSamples)
+{
+    int samplesIndex = 0;
+    QMutexLocker locker(&m_mutex);
+
+    // fill remainder of buffer and send it
+    if (m_sampleBufferIndex + nbSamples > m_packetSamples)
+    {
+        memcpy(&m_byteBuffer[m_sampleBufferIndex*m_sampleBytes],
+                &samples[samplesIndex*m_sampleBytes],
+                (m_packetSamples - m_sampleBufferIndex)*m_sampleBytes);
+
+        m_rtpSession.SendPacket((const void *) m_byteBuffer, (std::size_t) m_bufferSize);
+        nbSamples -= (m_packetSamples - m_sampleBufferIndex);
+        m_sampleBufferIndex = 0;
+    }
+
+    // send complete packets
+    while (nbSamples > m_packetSamples)
+    {
+        m_rtpSession.SendPacket((const void *) &samples[samplesIndex*m_sampleBytes], (std::size_t) m_bufferSize);
+        samplesIndex += m_packetSamples;
+        nbSamples -= m_packetSamples;
+    }
+
+    // copy remainder of input to buffer
+    memcpy(&m_byteBuffer[m_sampleBufferIndex*m_sampleBytes],
+            &samples[samplesIndex*m_sampleBytes],
+            nbSamples*m_sampleBytes);
+}
+

sdrbase/util/rtpsink.h

@@ -30,112 +30,40 @@
 #include "rtperrors.h"
 #include "rtplibraryversion.h"
 
-template<typename SampleType>
 class RTPSink
 {
 public:
-    RTPSink(const QString& address, uint16_t port, int sampleRate) :
+    typedef enum
-        m_destport(port),
-        m_sampleBuffer(0),
-        m_sampleBufferIndex(0),
-        m_mutex(QMutex::Recursive)
     {
-        m_destip = inet_addr(address.toStdString().c_str());
+        PayloadL16Mono,
-        m_destip = ntohl(m_destip);
+        PayloadL16Stereo,
-        m_rtpSessionParams.SetOwnTimestampUnit(1.0/(double) sampleRate);
+    } PayloadType;
-        m_rtpTransmissionParams.SetPortbase(8092); // FIXME: sort this out
 
-        int status = m_rtpSession.Create(m_rtpSessionParams, &m_rtpTransmissionParams);
+    RTPSink(const QString& address, uint16_t port, PayloadType payloadType = PayloadL16Mono);
+    ~RTPSink();
 
-        if (status < 0)
+    void setPayloadType(PayloadType payloadType);
-        {
-            qCritical("RTPSink::RTPSink: cannot create session: %s", jrtplib::RTPGetErrorString(status).c_str());
-            return;
-        }
 
-        status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
+    void setDestination(const QString& address, uint16_t port);
+    void deleteDestination(const QString& address, uint16_t port);
+    void addDestination(const QString& address, uint16_t port);
 
-        if (status < 0)
+    void write(uint8_t *sampleByte);
-        {
+    void write(uint8_t *sampleByte, int nbSamples);
-            qCritical("RTPSink::RTPSink: cannot set destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
-            return;
-        }
-
-        m_sampleBuffer = new SampleType[sampleRate]; // store 1 second
-    }
-
-    ~RTPSink()
-    {
-        jrtplib::RTPTime delay = jrtplib::RTPTime(10.0);
-        m_rtpSession.BYEDestroy(delay, "Time's up", 9);
-
-        if (m_sampleBuffer) { delete[] m_sampleBuffer; }
-    }
-
-    void setDestination(const QString& address, uint16_t port)
-    {
-        if (!m_sampleBuffer) { return; }
-
-        m_rtpSession.ClearDestinations();
-        m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
-        m_destip = inet_addr(address.toStdString().c_str());
-        m_destip = ntohl(m_destip);
-        m_destport = port;
-
-        int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(m_destip, m_destport));
-
-        if (status < 0) {
-            qCritical("RTPSink::setDestination: cannot set destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
-        }
-    }
-
-    void deleteDestination(const QString& address, uint16_t port)
-    {
-        uint32_t destip = inet_addr(address.toStdString().c_str());
-        destip = ntohl(m_destip);
-
-        int status =  m_rtpSession.DeleteDestination(jrtplib::RTPIPv4Address(destip, port));
-
-        if (status < 0) {
-            qCritical("RTPSink::deleteDestination: cannot delete destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
-        }
-    }
-
-    void addDestination(const QString& address, uint16_t port)
-    {
-        uint32_t destip = inet_addr(address.toStdString().c_str());
-        destip = ntohl(m_destip);
-
-        int status = m_rtpSession.AddDestination(jrtplib::RTPIPv4Address(destip, port));
-
-        if (status < 0) {
-            qCritical("RTPSink::addDestination: cannot add destination address: %s", jrtplib::RTPGetErrorString(status).c_str());
-        }
-    }
-
-    void setSampleRate(int sampleRate)
-    {
-        QMutexLocker locker(&m_mutex);
-        if (m_sampleBuffer) { delete[] m_sampleBuffer; }
-        m_sampleBuffer = new SampleType[sampleRate]; // store 1 second
-
-        int status = m_rtpSession.SetTimestampUnit(1.0 / (double) sampleRate);
-
-        if (status < 0)
-        {
-            qCritical("RTPSink::setSampleRate: cannot set timestamp unit: %s", jrtplib::RTPGetErrorString(status).c_str());
-            return;
-        }
-    }
 
 protected:
+    PayloadType m_payloadType;
+    int m_sampleRate;
+    int m_sampleBytes;
+    int m_packetSamples;
+    int m_bufferSize;
+    int m_sampleBufferIndex;
+    uint8_t *m_byteBuffer;
     uint32_t m_destip;
     uint16_t m_destport;
     jrtplib::RTPSession m_rtpSession;
     jrtplib::RTPSessionParams m_rtpSessionParams;
     jrtplib::RTPUDPv4TransmissionParams m_rtpTransmissionParams;
-    SampleType *m_sampleBuffer;
-    int m_sampleBufferIndex;
     QMutex m_mutex;
 };