- fix opus codec usage

src/engine/agent/Agent_Impl.cpp

@@ -208,10 +208,13 @@ void AgentImpl::processStart(JsonCpp::Value& /*request*/, JsonCpp::Value &answer
     PVariantMap priorityConfig = std::make_shared<VariantMap>();
     MT::CodecList& cl = mTerminal->codeclist();
     for (int i=0; i<cl.count(); i++)
-        if (cl.codecAt(i).payloadType() < 96)
+        priorityConfig->at(i) = i;
+
+    // Disable dynamic payload codec types - commented for now
+        /*if (cl.codecAt(i).payloadType() < 96)
             priorityConfig->at(i) = i;
         else
-            priorityConfig->at(i) = -1;
+            priorityConfig->at(i) = -1;*/
 
     config()[CONFIG_CODEC_PRIORITY] = priorityConfig;
 

src/engine/config.h

@@ -84,7 +84,7 @@
 
 // OPUS codec defines
 // #define USE_OPUS_CODEC
-#define MT_OPUS_CODEC_PT -1
+#define MT_OPUS_CODEC_PT 106
 
 // ILBC codec defines
 #define MT_ILBC20_PAYLOADTYPE -1

src/engine/endpoint/EP_AudioProvider.cpp

@@ -26,7 +26,7 @@ AudioProvider::AudioProvider(UserAgent& agent, MT::Terminal& terminal)
     if (mUserAgent.config().exists(CONFIG_CODEC_PRIORITY))
         mCodecPriority.setupFrom(mUserAgent.config()[CONFIG_CODEC_PRIORITY].asVMap());
     mSrtpSuite = SRTP_NONE;
-    setState((int)StreamState::SipRecv | (int)StreamState::SipSend | (int)StreamState::Receiving | (int)StreamState::Sending);
+    setStateImpl((int)StreamState::SipRecv | (int)StreamState::SipSend | (int)StreamState::Receiving | (int)StreamState::Sending);
 }
 
 AudioProvider::~AudioProvider()
@@ -182,6 +182,7 @@ void AudioProvider::sessionEstablished(int conntype)
     {
         RemoteCodec& rc = mAvailableCodecs.front();
         mActiveStream->setTransmittingCodec(*rc.mFactory, rc.mRemotePayloadType);
+        auto codec = dynamic_cast<MT::AudioStream*>(mActiveStream.get())->transmittingCodec();
         dynamic_cast<MT::AudioStream*>(mActiveStream.get())->setTelephoneCodec(mRemoteTelephoneCodec);
     }
 }
@@ -271,11 +272,10 @@ bool AudioProvider::processSdpOffer(const resip::SdpContents::Session::Medium& m
     return true;
 }
 
+
 void AudioProvider::setState(unsigned state)
 {
-    mState = state;
-    if (mActiveStream)
-        mActiveStream->setState(state);
+    setStateImpl(state);
 }
 
 unsigned AudioProvider::state()
@@ -381,3 +381,10 @@ void AudioProvider::setupMirror(bool enable)
     if (mActiveStream)
         mActiveStream->setupMirror(enable);
 }
+
+void AudioProvider::setStateImpl(unsigned int state) {
+    mState = state;
+    if (mActiveStream)
+        mActiveStream->setState(state);
+
+}

src/engine/endpoint/EP_AudioProvider.h

@@ -65,7 +65,8 @@ public:
   // myAnswer sets if the answer will be sent after.
   bool          processSdpOffer(const resip::SdpContents::Session::Medium& media, SdpDirection sdpDirection);
 
-  void          setState(unsigned state);
+
+  void          setState(unsigned state) override;
   unsigned      state();
   MT::Statistics  getStatistics();
   MT::PStream   activeStream();
@@ -112,6 +113,10 @@ protected:
   std::string createCryptoAttribute(SrtpSuite suite);
   SrtpSuite processCryptoAttribute(const resip::Data& value, ByteBuffer& key);
   void findRfc2833(const resip::SdpContents::Session::Medium::CodecContainer& codecs);
+
+  // Implements setState() logic. This allows to be called from constructor (it is not virtual function)
+  void setStateImpl(unsigned state);
+
 };
 
 #endif

src/engine/endpoint/EP_Session.cpp

@@ -726,7 +726,7 @@ void Session::buildSdp(resip::SdpContents &sdp, SdpDirection sdpDirection)
         if (mUserAgent->mConfig[CONFIG_MULTIPLEXING].asBool())
             media.addAttribute("rtcp-mux");
 
-        // Ask provider about specific information
+        // Ask provider about specific information - codecs are filled here
         provider.updateSdpOffer(media, sdpDirection);
 
         // Add ICE information

src/engine/media/MT_AudioCodec.cpp

@@ -402,6 +402,7 @@ PCodec OpusCodec::OpusFactory::create()
   result->applyParams(mParams);
   PCodec c(result);
   mCodecList.push_back(c);
+
   return c;
 }
 

src/engine/media/MT_AudioStream.cpp

@@ -110,6 +110,7 @@ void AudioStream::setDestination(const RtpPair<InternetAddress>& dest)
 void AudioStream::setTransmittingCodec(Codec::Factory& factory, int payloadType)  
 {
   ICELogInfo(<< "Selected codec " << factory.name() << "/" << factory.samplerate() << " for transmitting");
+
   Lock l(mMutex);
   mTransmittingCodec = factory.create();
   mTransmittingPayloadType = payloadType;
@@ -145,9 +146,11 @@ void AudioStream::addData(const void* buffer, int bytes)
   {
     Lock l(mMutex);
     codec = mTransmittingCodec.get();
-    if (!codec)
+    if (nullptr == codec) {
+      ICELogDebug(<< "No transmitting codec selected.");
       return;
     }
+  }
 
   // Resample
   unsigned dstlen = unsigned(float(codec->samplerate() / float(AUDIO_SAMPLERATE)) * bytes);
@@ -202,7 +205,7 @@ void AudioStream::addData(const void* buffer, int bytes)
   int packetTime = mPacketTime ? mPacketTime : codec->frameTime();
 
   // Make stereo version if required
-  for (int i=0; i<mCapturedAudio.filled() / mTransmittingCodec->pcmLength(); i++)
+  for (int i=0; i<mCapturedAudio.filled() / codec->pcmLength(); i++)
   {
     if (mSendingDump)
       mSendingDump->write((const char*)mCapturedAudio.data() + codec->pcmLength() * i, codec->pcmLength());
@@ -230,6 +233,7 @@ void AudioStream::addData(const void* buffer, int bytes)
       }
     }
   }
+  if (processed > 0)
     mCapturedAudio.erase(processed);
 }
 

src/engine/media/MT_AudioStream.h

@@ -64,7 +64,7 @@ namespace MT
     Audio::DataWindow mStereoCapturedAudio;
     char mIncomingPcmBuffer[AUDIO_MIC_BUFFER_SIZE]; // Temporary buffer to allow reading from file
     char mResampleBuffer[AUDIO_MIC_BUFFER_SIZE*8];  // Temporary buffer to hold data
-    char mStereoBuffer[AUDIO_MIC_BUFFER_SIZE*8];    // Temporary buffer to hold data converted to stereo
+    char mStereoBuffer[AUDIO_MIC_BUFFER_SIZE*16];   // Temporary buffer to hold data converted to stereo
     PCodec mTransmittingCodec;                      // Current encoding codec
     int mTransmittingPayloadType;                   // Payload type to mark outgoing packets
     int mPacketTime;                                // Required packet time

src/engine/media/MT_CodecList.cpp

@@ -128,12 +128,12 @@ CodecList::CodecList(const Settings& settings)
 
   //mFactoryList.push_back(new IsacCodec::IsacFactory16K(mSettings.mIsac16KPayloadType));
   //mFactoryList.push_back(new IlbcCodec::IlbcFactory(mSettings.mIlbc20PayloadType, mSettings.mIlbc30PayloadType));
-  mFactoryList.push_back(new G711Codec::AlawFactory());
-  mFactoryList.push_back(new G711Codec::UlawFactory());
+  // mFactoryList.push_back(new G711Codec::AlawFactory());
+  // mFactoryList.push_back(new G711Codec::UlawFactory());
 
-  mFactoryList.push_back(new GsmCodec::GsmFactory(mSettings.mGsmFrPayloadLength == 32 ? GsmCodec::Type::Bytes_32 : GsmCodec::Type::Bytes_33, mSettings.mGsmFrPayloadType));
-  mFactoryList.push_back(new G722Codec::G722Factory());
-  mFactoryList.push_back(new G729Codec::G729Factory());
+  // mFactoryList.push_back(new GsmCodec::GsmFactory(mSettings.mGsmFrPayloadLength == 32 ? GsmCodec::Type::Bytes_32 : GsmCodec::Type::Bytes_33, mSettings.mGsmFrPayloadType));
+  // mFactoryList.push_back(new G722Codec::G722Factory());
+  // mFactoryList.push_back(new G729Codec::G729Factory());
 #ifndef TARGET_ANDROID
   mFactoryList.push_back(new GsmHrCodec::GsmHrFactory(mSettings.mGsmHrPayloadType));
 #endif
@@ -231,7 +231,7 @@ void CodecListPriority::setupFrom(PVariantMap vmap)
     {
       Item item;
       item.mCodecIndex = i;
-      item.mPriority = vmap->exists(i) ? vmap->at(i).asInt() : -1;
+      item.mPriority = vmap->exists(i) ? vmap->at(i).asInt() : 1000; // Non listed codecs will get lower priority
       mPriorityList.push_back(item);
     }