- more cleanups - use uuid library on the rtphone level + formatting fixed

src/engine/helper/CMakeLists.txt

@@ -15,3 +15,4 @@ set_property(TARGET helper_lib PROPERTY MSVC_RUNTIME_LIBRARY "MultiThreaded$<$<C
 # Private include directories
 target_include_directories(helper_lib PUBLIC ../../libs/ ../../engine ../ .)
 target_compile_definitions(helper_lib PRIVATE -D_CRT_SECURE_NO_WARNINGS -D_UNICODE)
+target_link_libraries(helper_lib PUBLIC uuid)

src/engine/media/MT_AudioReceiver.cpp

@@ -331,6 +331,7 @@ AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics
     mResampler48.start(AUDIO_CHANNELS, 48000, AUDIO_SAMPLERATE);
 
     // Init codecs
+    mCodecList.setSettings(settings);
     mCodecList.fillCodecMap(mCodecMap);
 
 #if defined(DUMP_DECODED)
@@ -442,14 +443,14 @@ bool AudioReceiver::add(const std::shared_ptr<jrtplib::RTPPacket>& p, Codec** co
     if (payloadLength >= 1 && payloadLength <= 6 && (ptype == 0 || ptype == 8))
         time_length = mLastPacketTimeLength ? mLastPacketTimeLength : 20;
     else
-        // Check if packet is too short from time length side
-        if (time_length < 2)
-        {
-            // It will cause statistics to report about bad RTP packet
-            // I have to replay last packet payload here to avoid report about lost packet
-            mBuffer.add(p, time_length, codecIter->second->samplerate());
-            return false;
-        }
+    // Check if packet is too short from time length side
+    if (time_length < 2)
+    {
+        // It will cause statistics to report about bad RTP packet
+        // I have to replay last packet payload here to avoid report about lost packet
+        mBuffer.add(p, time_length, codecIter->second->samplerate());
+        return false;
+    }
 
     // Queue packet to buffer
     auto packet = mBuffer.add(p, time_length, codecIter->second->samplerate()).get();

src/engine/media/MT_Statistics.cpp

@@ -1,4 +1,5 @@
 #include <math.h>
+#include <iostream>
 
 #include "MT_Statistics.h"
 #include "audio/Audio_Interface.h"
@@ -55,8 +56,11 @@ void JitterStatistics::process(jrtplib::RTPPacket* packet, int rate)
         mReceiveTime = receiveTime;
         mReceiveTimestamp = timestamp;
 
-        // And mJitter are in seconds again
-        mJitter.process(mLastJitter.value() / float(rate));
+        // And mJitter are in milliseconds again
+        float jitter_s = mLastJitter.value() / (float(rate));
+        // std::cout << "Jitter (in seconds): " << std::dec << jitter_s << std::endl;
+
+        mJitter.process(jitter_s);
     }
 }