diff --git a/.vscode/c_cpp_properties.json b/.vscode/c_cpp_properties.json
index acfe4b839df398bb75e89f155451e99ba68ae54f..7ec2fc7f6c9d200b77751e445160aee544b438e6 100644
--- a/.vscode/c_cpp_properties.json
+++ b/.vscode/c_cpp_properties.json
@@ -108,7 +108,8 @@
             "defines": [
                 "LINUX",
                 "__DEBUG__=1",
-                "INCLUDE_DSPE_EXAMPLES"
+                "INCLUDE_DSPE_EXAMPLES",
+                "AUDIO_DEBUG_MESSAGES_ON"
             ],
             "compilerPath": "/usr/bin/gcc",
             "cStandard": "c11",
diff --git a/CHANGELOG b/CHANGELOG
index cadb94adbd80910013d4d785f58488dbc9cbfa47..7d8878a218bec9c6b109b8dcabbb7680dccea74a 100644
--- a/CHANGELOG
+++ b/CHANGELOG
@@ -1,4 +1,5 @@
-TODO::
+TODO:
+  - test socket support on Linux (test examples as there are have been noticed some problems with bind)
   ? DSP::Clock_ptr => migrate to std::shared_ptr
   !!! Add TELEsound and PiAPS_sound projects to GitLab and start addaptation to new library version
 
diff --git a/src/Makefile.linux b/src/Makefile.linux
index 2443d6c0b481b9c986ba6bb69521a7aefec7fcbc..695337de853a4627f5bcd9ab2ff8d2d82d23977b 100644
--- a/src/Makefile.linux
+++ b/src/Makefile.linux
@@ -24,7 +24,7 @@ INCLUDES_RLS := -I./src/include -I./src/include/rls
 LIBS := -lasound
 
 # \TODO is DEVCPP ok or is it unnecessary
-DFLAGS         = -DLINUX -DDEVCPP -DINCLUDE_DSPE_EXAMPLES
+DFLAGS         = -DLINUX -DDEVCPP -DINCLUDE_DSPE_EXAMPLES -DAUDIO_DEBUG_MESSAGES_ON
 CFLAGS_release = $(comflag) -std=c++0x -O3 -Wall -c -fmessage-length=0 -fno-strict-aliasing 
 CFLAGS_debug   = $(comflag) -std=c++0x -O0 -g3 -Wall -c -fmessage-length=0 -W -Wshadow -Wconversion -fstrict-aliasing -fmax-errors=5
 # -U__STRICT_ANSI__ jest potrzebne do kompilacji debug_new.cpp, je�eli pomin�� ten plik to mo�na r�wnie� wyrzuci� t� opcj�
diff --git a/src/cpp/ALSA_support.cpp b/src/cpp/ALSA_support.cpp
index 5ec529179de7027d152a98cc9969b0c6b676b267..73cf4072a4896cdf25809c2d6d4d928a5a2c042e 100644
--- a/src/cpp/ALSA_support.cpp
+++ b/src/cpp/ALSA_support.cpp
@@ -174,9 +174,7 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
 
   if (rc < 0)
   {
-    #ifdef AUDIO_DEBUG_MESSAGES_ON
-      DSP::log << "Unable to open pcm device: " << snd_strerror(rc) << endl;
-    #endif // AUDIO_DEBUG_MESSAGES_ON
+    DSP::log << "Unable to open pcm device: " << snd_strerror(rc) << endl;
 
     return -1;
   }
@@ -210,19 +208,19 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
 
   rc = snd_pcm_hw_params_set_buffer_size(alsa_handle, params, DSP::NoOfAudioOutputBuffers*audio_inbuffer_size_in_frames);
     
-  snd_pcm_uframes_t tmp_frames = audio_inbuffer_size_in_frames;
+  snd_pcm_uframes_t requested_audio_inbuffer_size_in_frames = audio_inbuffer_size_in_frames;
 
-  #ifdef AUDIO_DEBUG_MESSAGES_ON
+  if (rc < 0) {
     DSP::log << "Buffer size set with error code: " << rc << endl;
-  #endif // AUDIO_DEBUG_MESSAGES_ON
+  }
 
   /*! Set period size to desired number of frames. */
   snd_pcm_hw_params_set_period_size_near(alsa_handle, params, &audio_inbuffer_size_in_frames, &dir);
 
   #ifdef AUDIO_DEBUG_MESSAGES_ON
-    if (audio_inbuffer_size_in_frames != tmp_frames)
+    if (audio_inbuffer_size_in_frames != requested_audio_inbuffer_size_in_frames)
     {
-      DSP::log << "Current frames value should be equal: " << tmp_frames << endl;
+      DSP::log << "Current frames value should be equal: " << requested_audio_inbuffer_size_in_frames << endl;
       DSP::log << "Frames is not equal to tmp_frames! Frames: " << audio_inbuffer_size_in_frames << endl;    
     }
 
@@ -235,9 +233,8 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
 
   if (rc < 0)
   {
-    #ifdef AUDIO_DEBUG_MESSAGES_ON
-      DSP::log << "Unable to set hw parameters: " << snd_strerror(rc) << endl;
-    #endif // AUDIO_DEBUG_MESSAGES_ON
+    DSP::log << "Unable to set hw parameters: " << snd_strerror(rc) << endl;
+    DSP::log << "Closing ALSA device" << endl;
 
     close_alsa_device();
       
@@ -419,9 +416,7 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
       break;
 
     default:
-      #ifdef AUDIO_DEBUG_MESSAGES_ON
-        DSP::log << "Unsupported no of bytes in channel" << endl;
-      #endif // AUDIO_DEBUG_MESSAGES_ON
+      DSP::log << "Unsupported no of bytes in channel" << endl;
       
       return -1;
       break;