- fixes

src/engine/agent/Agent_Impl.cpp

@@ -5,6 +5,7 @@
 #include "helper/HL_VariantMap.h"
 #include "helper/HL_CsvReader.h"
 #include "helper/HL_Base64.h"
+#include "media/MT_CodecList.h"
 #include <fstream>
 
 
@@ -210,7 +211,8 @@ void AgentImpl::processStart(JsonCpp::Value& request, JsonCpp::Value &answer)
     SocketHeap::instance().start();
 
     // Initialize terminal
-    MT::CodecList::Settings settings;
+    auto settings = MT::CodecList::Settings::getClientSettings();
+
     mTerminal = std::make_shared<MT::Terminal>(settings);
 
     // Enable/disable codecs
@@ -220,10 +222,10 @@ void AgentImpl::processStart(JsonCpp::Value& request, JsonCpp::Value &answer)
         priorityConfig->at(i) = i;
 
     // Disable dynamic payload codec types - commented for now
-        /*if (cl.codecAt(i).payloadType() < 96)
+    // if (cl.codecAt(i).payloadType() < 96)
-            priorityConfig->at(i) = i;
+    //    priorityConfig->at(i) = i;
-        else
+    // else
-            priorityConfig->at(i) = -1;*/
+    //    priorityConfig->at(i) = -1;
 
     config()[CONFIG_CODEC_PRIORITY] = priorityConfig;
 

src/engine/media/MT_AudioReceiver.cpp

@@ -423,13 +423,13 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** de
         payloadLength = p->GetPayloadLength(),
         ptype = p->GetPayloadType();
 
-    // ICELogInfo(<< "Adding packet No " << p->GetSequenceNumber());
+    ICELogInfo(<< "Adding packet No " << p->GetSequenceNumber());
     // Increase codec counter
     mStat.mCodecCount[ptype]++;
 
     // Check if codec can be handled
     Codec* codec = nullptr;
-    CodecMap::iterator codecIter = mCodecMap.find(ptype);
+    auto codecIter = mCodecMap.find(ptype);
     if (codecIter == mCodecMap.end())
     {
         time_length = 10;
@@ -618,7 +618,17 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacket(const RtpBuffer::ResultL
 
         // Find codec by payload type
         int ptype = p->rtp()->GetPayloadType();
-        mCodec = mCodecMap[ptype];
+
+        // Look into mCodecMap if exists
+        auto codecIter = mCodecMap.find(ptype);
+        if (codecIter == mCodecMap.end())
+            return  {};
+
+
+        if (!codecIter->second)
+            codecIter->second = mCodecList.createCodecByPayloadType(ptype);
+
+        mCodec = codecIter->second;
         if (mCodec)
         {
             if (rate)
@@ -695,7 +705,7 @@ AudioReceiver::DecodeResult AudioReceiver::decodePacket(const RtpBuffer::ResultL
 
 AudioReceiver::DecodeResult AudioReceiver::decodeNone(Audio::DataWindow& output, DecodeOptions options)
 {
-    ICELogDebug(<< "No packet available in jitter buffer");
+    // ICELogDebug(<< "No packet available in jitter buffer");
     mFailedCount++;
     return DecodeResult_Skip;
 }

src/engine/media/MT_AudioStream.cpp

@@ -225,9 +225,9 @@ void AudioStream::addData(const void* buffer, int bytes)
             if (packetTime <= encodedTime)
             {
                 // Time to send packet
-                ICELogMedia(<< "Sending RTP packet pt = " << mTransmittingPayloadType << ", plength = " << (int)mEncodedAudio.size());
+                ICELogMedia(<< "Sending RTP packet pt = " << mTransmittingPayloadType << ", plength = " << (int)mEncodedAudio.size() << " to ");
                 mRtpSession.SendPacketEx(mEncodedAudio.data(), mEncodedAudio.size(), mTransmittingPayloadType, false,
-                                         packetTime * codec->samplerate()/1000, 0, NULL, 0);
+                                         packetTime * codec->samplerate()/1000, 0, nullptr, 0);
                 mEncodedAudio.clear();
                 encodedTime = 0;
             }
@@ -365,9 +365,11 @@ void AudioStream::dataArrived(PDatagramSocket s, const void* buffer, int length,
 
             // Find right handler for rtp stream
             SingleAudioStream* rtpStream = nullptr;
-            AudioStreamMap::iterator streamIter = mStreamMap.find(packet->GetSSRC());
+            auto streamIter = mStreamMap.find(packet->GetSSRC());
-            if (streamIter == mStreamMap.end())
+            if (streamIter == mStreamMap.end()) {
-                mStreamMap[packet->GetSSRC()] = rtpStream = new SingleAudioStream(mCodecSettings, mStat);
+                rtpStream = new SingleAudioStream(mCodecSettings, mStat);
+                mStreamMap.insert({packet->GetSSRC(), rtpStream});
+            }
             else
                 rtpStream = streamIter->second;
 

src/engine/media/MT_AudioStream.h

@@ -62,16 +62,16 @@ public:
 protected:
     Audio::DataWindow mCapturedAudio;     // Data from microphone
     Audio::DataWindow mStereoCapturedAudio;
-    char mIncomingPcmBuffer[AUDIO_MIC_BUFFER_SIZE]; // Temporary buffer to allow reading from file
+    char mIncomingPcmBuffer[AUDIO_MIC_BUFFER_SIZE] = {0}; // Temporary buffer to allow reading from file
-    char mResampleBuffer[AUDIO_MIC_BUFFER_SIZE*8];  // Temporary buffer to hold data
+    char mResampleBuffer[AUDIO_MIC_BUFFER_SIZE*8] = {0};  // Temporary buffer to hold data
-    char mStereoBuffer[AUDIO_MIC_BUFFER_SIZE*16];   // Temporary buffer to hold data converted to stereo
+    char mStereoBuffer[AUDIO_MIC_BUFFER_SIZE*16] = {0};   // Temporary buffer to hold data converted to stereo
     PCodec mTransmittingCodec;                      // Current encoding codec
-    int mTransmittingPayloadType;                   // Payload type to mark outgoing packets
+    int mTransmittingPayloadType = -1;              // Payload type to mark outgoing packets
-    int mPacketTime;                                // Required packet time
+    int mPacketTime = 0;                            // Required packet time
     char mFrameBuffer[MT_MAXAUDIOFRAME];            // Temporary buffer to hold results of encoder
     ByteBuffer mEncodedAudio;                       // Encoded frame(s)
-    int mEncodedTime;                               // Time length of encoded audio
+    int mEncodedTime = 0;                           // Time length of encoded audio
-    const CodecList::Settings& mCodecSettings;      // Configuration for stream
+    CodecList::Settings mCodecSettings;             // Configuration for stream
     Mutex mMutex;                      	            // Mutex
     int mRemoteTelephoneCodec;                      // Payload for remote telephone codec
     jrtplib::RTPSession mRtpSession;                // Rtp session
@@ -87,8 +87,8 @@ protected:
                       mCaptureResampler32,
                       mCaptureResampler48;
     DtmfContext mDtmfContext;
-    char mReceiveBuffer[MAX_VALID_UDPPACKET_SIZE],
+    char mReceiveBuffer[MAX_VALID_UDPPACKET_SIZE] = {0},
-         mSrtpDecodeBuffer[MAX_VALID_UDPPACKET_SIZE];
+         mSrtpDecodeBuffer[MAX_VALID_UDPPACKET_SIZE] = {0};
 
     struct
     {
@@ -106,7 +106,7 @@ protected:
 
     Statistics* mFinalStatistics = nullptr;
 
-    bool decryptSrtp(void* data, int* len);
+    // bool decryptSrtp(void* data, int* len);
 };
 };
 

src/engine/media/MT_Box.cpp

@@ -40,7 +40,7 @@ PStream Terminal::createStream(int type, VariantMap& /*config*/)
   switch (type)
   {
   case Stream::Audio:
-    result = std::make_shared<AudioStream>(MT::CodecList::Settings::DefaultSettings);
+    result = std::make_shared<AudioStream>(MT::CodecList::Settings::getClientSettings());
     mAudioList.add(result);
     break;
 

src/engine/media/MT_CodecList.cpp

@@ -25,6 +25,17 @@ using namespace MT;
 
 using strx = strx;
 
+CodecList::Settings CodecList::Settings::getAnalyzerSettings()
+{
+    return {};
+}
+
+CodecList::Settings CodecList::Settings::getClientSettings()
+{
+    Settings r;
+    r.mOpusSpec.push_back(Settings::OpusSpec(MT_OPUS_CODEC_PT, 48000, 2));
+    return r;
+}
 
 bool CodecList::Settings::contains(int ptype) const
 {
@@ -223,7 +234,7 @@ static int findOctetMode(const char* line)
 
 CodecList::Settings CodecList::Settings::parseSdp(const std::list<resip::Codec>& codeclist)
 {
-    CodecList::Settings r{DefaultSettings};
+    CodecList::Settings r;
 
     for (auto& c: codeclist)
     {
@@ -235,41 +246,27 @@ CodecList::Settings CodecList::Settings::parseSdp(const std::list<resip::Codec>&
         auto params = c.parameters();
 
         // Dynamic payload type codecs only - ISAC / iLBC / Speex / etc.
-        if (codec_name == "OPUS")
+        if (codec_name == "OPUS") {
-        {
             // Check the parameters
             int channels = strx::toInt(enc_params.c_str(), 1);
             r.mOpusSpec.push_back({ptype, samplerate, channels});
-        }
+        } else if (codec_name == "AMR-WB") {
-        else
-        if (codec_name == "AMR-WB")
-        {
             int octet_mode = findOctetMode(params.c_str());
-            if (octet_mode != -1)
+            if (octet_mode != -1) {
-            {
                 if (octet_mode == 0)
                     r.mAmrWbPayloadType.insert(ptype);
-                else
+                else if (octet_mode == 1)
-                if (octet_mode == 1)
                     r.mAmrWbOctetPayloadType.insert(ptype);
             }
-        }
+        } else if (codec_name == "AMR" || codec_name == "AMR-NB") {
-        else
-        if (codec_name == "AMR" || codec_name == "AMR-NB")
-        {
             int octet_mode = findOctetMode(params.c_str());
-            if (octet_mode != -1)
+            if (octet_mode != -1) {
-            {
                 if (octet_mode == 0)
                     r.mAmrNbPayloadType.insert(ptype);
-                else
+                else if (octet_mode == 1)
-                if (octet_mode == 1)
                     r.mAmrNbOctetPayloadType.insert(ptype);
             }
-        }
+        } else if (codec_name == "EVS") {
-        else
-        if (codec_name == "EVS")
-        {
             r.mEvsSpec.push_back({ptype});
         }
     }
@@ -314,7 +311,6 @@ bool CodecList::Settings::operator == (const Settings& rhs) const
 
 
 // ----------------------------------------
-CodecList::Settings CodecList::Settings::DefaultSettings;
 
 CodecList::CodecList(const Settings& settings)
     :mSettings(settings)
@@ -405,8 +401,8 @@ void CodecList::fillCodecMap(CodecMap& cm)
     for (auto& factory: mFactoryList)
     {
         // Create codec here. Although they are not needed right now - they can be needed to find codec's info.
-        // PCodec c = factory->create();
+        PCodec c = factory->create();
-        cm.insert({factory->payloadType(), PCodec()});
+        cm.insert({factory->payloadType(), c});
     }
 }
 
@@ -438,7 +434,7 @@ bool CodecListPriority::compare(const Item& item1, const Item& item2)
 
 void CodecListPriority::setupFrom(PVariantMap vmap)
 {
-    CodecList::Settings settings;
+    auto settings = CodecList::Settings::getClientSettings();
     CodecList cl(settings);
     //mPriorityList.resize(cl.count());
     bool emptyVmap = vmap ? vmap->empty() : true;

src/engine/media/MT_CodecList.h

@@ -101,11 +101,12 @@ public:
         std::string toString() const;
         void        clear();
 
-        static Settings DefaultSettings;
-
         static Settings parseSdp(const std::list<resip::Codec>& codeclist);
 
         bool operator == (const Settings& rhs) const;
+
+        static Settings getClientSettings();
+        static Settings getAnalyzerSettings();
     };
 
     CodecList(const Settings& settings);

src/engine/media/MT_NativeRtpSender.cpp

@@ -1,15 +1,17 @@
-/* Copyright(C) 2007-2019 VoIP objects (voipobjects.com)
+/* Copyright(C) 2007-2026 VoIP objects (voipobjects.com)
  * This Source Code Form is subject to the terms of the Mozilla Public
  * License, v. 2.0. If a copy of the MPL was not distributed with this
  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "MT_NativeRtpSender.h"
+#include "ICELog.h"
 #include <assert.h>
 
+#define LOG_SUBSYSTEM "MT"
 using namespace MT;
 
 NativeRtpSender::NativeRtpSender(Statistics& stat)
-:mStat(stat), mSrtpSession(nullptr)
+    :mStat(stat), mSrtpSession(nullptr)
 {
 }
 
@@ -41,6 +43,7 @@ bool NativeRtpSender::SendRTP(const void *data, size_t len)
         }
     }
 
+    ICELogInfo(<< "Sending " << sendLength <<" bytes to " << mTarget.mRtp.toBriefStdString());
     mSocket.mRtp->sendDatagram(mTarget.mRtp, mSendBuffer, sendLength);
     mStat.mSentRtp++;
     mStat.mSent += len;

src/engine/media/MT_NativeRtpSender.h

@@ -18,9 +18,9 @@
 
 namespace MT
 {
-  class NativeRtpSender: public jrtplib::RTPExternalSender
+class NativeRtpSender: public jrtplib::RTPExternalSender
-  {
+{
-  public:
+public:
     NativeRtpSender(Statistics& stat);
     ~NativeRtpSender();
     
@@ -46,7 +46,7 @@ namespace MT
     void setSrtpSession(SrtpSession* srtp);
     SrtpSession* srtpSession();
 
-  protected:
+protected:
     RtpPair<PDatagramSocket> mSocket;
     RtpPair<InternetAddress> mTarget;
     Statistics& mStat;
@@ -55,7 +55,7 @@ namespace MT
 #endif
     SrtpSession* mSrtpSession;
     char mSendBuffer[MAX_VALID_UDPPACKET_SIZE];
-  };
+};
 }
 
 #endif

src/engine/media/MT_SingleAudioStream.cpp

@@ -39,7 +39,7 @@ void SingleAudioStream::copyPcmTo(Audio::DataWindow& output, int needed)
             break;
     }
 
-    if (output.filled() < needed)
+    // if (output.filled() < needed)
-        ICELogError(<< "Not enough data for speaker's mixer");
+    //    ICELogError(<< "Not enough data for speaker's mixer");
 }