reconnect functionality added

StreamRecorder.h

@@ -11,62 +11,83 @@
 #ifndef STREAMRECORDER_H
 #define STREAMRECORDER_H
 
-// your public header include
 //------------------------------------------------------------------------------
+
+/** Your public header include */
+
 #include <gst/gst.h>
 #include <jmorecfg.h>
+
 //------------------------------------------------------------------------------
+
 #define STREAMRECORDER_SAMPLERATE 44100
 #define READSIZE 1152 //For MPEG1, frame_size = 1152 samples/frame
 #define STREAMRECORDER_BYTESPERSAMPLE 2
-// the declaration of your class...
+
+#define ERROR_MSG_SIZE 50
+#define DST_URI_SIZE 80
+
 //------------------------------------------------------------------------------
+
+/** Class declaration */
+
 class StreamRecorder
 {
-    private:
+	private:
+    
+        unsigned char* audioBuffer;
+        unsigned char* audioBufferPosition;
+    
+        char errorMessage[ERROR_MSG_SIZE];
+        char pluginUri[DST_URI_SIZE];
+	
+	    unsigned int nBytes;
+	    unsigned int bufferSize;
 
-        unsigned int nBytes;
-	unsigned int bufferSize;
-	unsigned char* audioBuffer;
-	unsigned char* audioBufferPosition;
-     	int recordTime;
+	    int recordTime;
+        int audioFileDuration;
+    
+        /** Audio filename */
+        long int timestamp = 0;
+    
         bool isConnectionLost;
     
-        char * pluginUri;
-        //char* sourceName;
-     	GstElement* streamSrc;
-     	GstElement* audioConvert;
-     	//GstElement* audioResample;
-     	GstElement* filterCaps;
-     	GstElement* queue0;
-     	GstElement* queue1;
-     	GstElement* filter;
-     	GstElement* fakeSink;
-     	//GstElement* audioSink;
-     	GstElement* mainPipeline;
-     	//GstElement* tempBin;
+	    //char* sourceName;
+        //GstElement* audioResample;
+        //GstElement* tempBin;
+        //GstElement* audioSink;
     
-	int createMainPipeline();
-	int connect(const char *uri);
-	int disconnect();
+	    GstElement* streamSrc;
+	    GstElement* audioConvert;
+        GstElement* filterCaps;
+        GstElement* queue0;
+        GstElement* queue1;
+        GstElement* filter;
+        GstElement* fakeSink;
+        GstElement* mainPipeline;
     
-        // callbacks
+        int createMainPipeline();
+        int connect();
+        int disconnect();
+    
+        /** add data to buffer */
+        int addToBuffer(unsigned char* data, int length);
+        int compressBuffer();
+        
+        /** plugin's callbacks */
         static void srcNewPad_callback(GstElement *element, GstPad *pad, void *data);
         static int bus_callback(GstBus *bus, GstMessage *message, void *data);
-        static int filter_handoff_callback(GstElement* filter, GstBuffer* buffer, void* user_data);
-		
-        // add data to buffer
-        int addToBuffer(unsigned char* data, int length);
-	int compressBuffer();
+        static int filter_handoff_callback(GstElement *filter, GstBuffer* buffer, void* user_data);
+        
+        /** Save audio*/
+        static void saveBuffer(void *user_data);
+        
+        /** Restart the pipeline */
+        static gboolean reconnectURIStream(void* data);
     
-        // Save information when connection fails
-	static void savePartialBuffer(void *user_data);
-	    
-        // Restart the pipeline
-	static gboolean reconnectURIStream(void *data);
+	public:
     
-    public:	    
-	StreamRecorder(const char* source, int time);
+	    StreamRecorder(const char* source, int time);
 };
 //------------------------------------------------------------------------------
 #endif