Added define for audio subsampling.

Author: AnBenLa

CMakeLists.txt

@@ -1,5 +1,5 @@
 cmake_minimum_required(VERSION 3.16)
-project(RecordingPlugin VERSION "0.8.4")
+project(RecordingPlugin VERSION "0.8.5")
 
 set(CMAKE_CXX_STANDARD 17)
 
@@ -13,9 +13,9 @@ set(UtilsDir Utils)
 configure_file(Config.h.in ${CMAKE_CURRENT_SOURCE_DIR}/${SrcDir}/Config.h @ONLY)
 
 if(ANDROID_BUILD)
-    set( UNITY_PLUGINS_DIR "C:/Users/Anton-Lammert/Desktop/ImmersiveStudyAnalyzer/Assets/Plugins/Android/ARM64" )
+    set( UNITY_PLUGINS_DIR "C:/Users/Anton-Lammert/Desktop/Immersive-Study-Analyzer/Assets/Plugins/Android/ARM64" )
 else()
-    set( UNITY_PLUGINS_DIR "C:/Users/Anton-Lammert/Desktop/ImmersiveStudyAnalyzer/Assets/Plugins/Windows" )
+    set( UNITY_PLUGINS_DIR "C:/Users/Anton-Lammert/Desktop/Immersive-Study-Analyzer/Assets/Plugins/Windows" )
 endif()
 
 if(NOT WORKFLOW)

Config.h.in

@@ -10,6 +10,7 @@
 #define PROJECT_VER_MINOR "@PROJECT_VERSION_MINOR@"
 #define PROJECT_VER_PATCH "@PROJECT_VERSION_PATCH@"
 #define COMPRESSION 0
+#define SUBSAMPLE_RECORDED_AUDIO 0
 #define COMPRESSION_LEVEL 5 // 1 (low) to 9 (high but slow)
 #define MULTI_THREADED 1
 const size_t HEADER_SIZE = sizeof(uint32_t); // size of compressed data block header

Src/Config.h

@@ -5,11 +5,12 @@
 #include <stdint.h>
 
 #define PROJECT_NAME "RecordingPlugin"
-#define PROJECT_VER  "0.8.4"
+#define PROJECT_VER  "0.8.5"
 #define PROJECT_VER_MAJOR "0"
 #define PROJECT_VER_MINOR "8"
-#define PROJECT_VER_PATCH "4"
+#define PROJECT_VER_PATCH "5"
 #define COMPRESSION 0
+#define SUBSAMPLE_RECORDED_AUDIO 0
 #define COMPRESSION_LEVEL 5 // 1 (low) to 9 (high but slow)
 #define MULTI_THREADED 1
 const size_t HEADER_SIZE = sizeof(uint32_t); // size of compressed data block header

Src/Recording/Generic/GenericData.h

@@ -60,7 +60,7 @@ struct GenericDTO{
     char c[char_c];
 
     std::string static dto_layout() {
-        return "genericDTO: int id;float t;int i[10];float f[10];char c[10];int layer = 0;";
+        return "genericDTO: int id;float t;int i[10];float f[10];char c[10];";
     }
 };
 

Src/Recording/Recorder.cpp

@@ -504,7 +504,7 @@ Recorder::record_sound(float *sound_data, int sound_data_length, int sampling_ra
                 Debug::Log("Waiting until the current sound data is written to file before recording new data");
             i++;
         }
-
+#ifdef SUBSAMPLE_RECORDED_AUDIO
         int subsample_factor = 2;
         bool subsampled = false;
         if(sampling_rate % subsample_factor == 0 && sampling_rate/subsample_factor >= 16000){
@@ -522,6 +522,7 @@ Recorder::record_sound(float *sound_data, int sound_data_length, int sampling_ra
 
         if(debug)
             Debug::Log("Subsampled audio source: " + std::to_string(sound_id));
+# endif
 
         if(last_sound_data.count(sound_id) == 0)
             last_sound_data[sound_id] = sound_dto;

Src/Recording/Sound/SoundData.h

@@ -60,7 +60,7 @@ struct SoundDTO{
     float s_d[4800];
 
     std::string static dto_layout(){
-        return "SoundDTO: int id;float t;float s_t;float e_t;int s_n;int s_i;int s_r;int c_n;int c_go_id;float s_d[4800];int layer = 0;";
+        return "SoundDTO: int id;float t;float s_t;float e_t;int s_n;int s_i;int s_r;int c_n;int c_go_id;float s_d[4800];";
     }
 };
 

Src/Recording/Transform/TransformData.h

@@ -91,7 +91,7 @@ struct TransformDTO{
     }
 
     std::string static dto_layout(){
-        return "TransformDTO: int id;float t;float lp[3];float lr[4];float ls[3];float gp[3];float gr[4];float gs[3];int act;int p_id;int layer = 0;";
+        return "TransformDTO: int id;float t;float lp[3];float lr[4];float ls[3];float gp[3];float gr[4];float gs[3];int act;int p_id;";
     }
 };
 

Utils/Utils.cpp

@@ -194,7 +194,7 @@ void Utils::export_generic_data_to_CSV(const std::string &generic_file_path) {
 
 void Utils::export_sound_data_to_WAV(std::string const& sound_file_path) {
     Debug::Log("Writing sound data to WAV file");
-
+    Debug::Log("Reading sound data from file: " + sound_file_path);
     std::string sound_data_file = sound_file_path;
     //std::set<int> sound_ids = current_recording.meta_information.get_sound_ids();