Commit 01b8230d authored by d.basulto's avatar d.basulto

reconnect functionality added

parent c9ad4f92
This diff is collapsed.
...@@ -11,61 +11,82 @@ ...@@ -11,61 +11,82 @@
#ifndef STREAMRECORDER_H #ifndef STREAMRECORDER_H
#define STREAMRECORDER_H #define STREAMRECORDER_H
// your public header include
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** Your public header include */
#include <gst/gst.h> #include <gst/gst.h>
#include <jmorecfg.h> #include <jmorecfg.h>
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
#define STREAMRECORDER_SAMPLERATE 44100 #define STREAMRECORDER_SAMPLERATE 44100
#define READSIZE 1152 //For MPEG1, frame_size = 1152 samples/frame #define READSIZE 1152 //For MPEG1, frame_size = 1152 samples/frame
#define STREAMRECORDER_BYTESPERSAMPLE 2 #define STREAMRECORDER_BYTESPERSAMPLE 2
// the declaration of your class...
#define ERROR_MSG_SIZE 50
#define DST_URI_SIZE 80
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
/** Class declaration */
class StreamRecorder class StreamRecorder
{ {
private: private:
unsigned int nBytes;
unsigned int bufferSize;
unsigned char* audioBuffer; unsigned char* audioBuffer;
unsigned char* audioBufferPosition; unsigned char* audioBufferPosition;
char errorMessage[ERROR_MSG_SIZE];
char pluginUri[DST_URI_SIZE];
unsigned int nBytes;
unsigned int bufferSize;
int recordTime; int recordTime;
int audioFileDuration;
/** Audio filename */
long int timestamp = 0;
bool isConnectionLost; bool isConnectionLost;
char * pluginUri;
//char* sourceName; //char* sourceName;
//GstElement* audioResample;
//GstElement* tempBin;
//GstElement* audioSink;
GstElement* streamSrc; GstElement* streamSrc;
GstElement* audioConvert; GstElement* audioConvert;
//GstElement* audioResample;
GstElement* filterCaps; GstElement* filterCaps;
GstElement* queue0; GstElement* queue0;
GstElement* queue1; GstElement* queue1;
GstElement* filter; GstElement* filter;
GstElement* fakeSink; GstElement* fakeSink;
//GstElement* audioSink;
GstElement* mainPipeline; GstElement* mainPipeline;
//GstElement* tempBin;
int createMainPipeline(); int createMainPipeline();
int connect(const char *uri); int connect();
int disconnect(); int disconnect();
// callbacks /** add data to buffer */
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 addToBuffer(unsigned char* data, int length);
int compressBuffer(); int compressBuffer();
// Save information when connection fails /** plugin's callbacks */
static void savePartialBuffer(void *user_data); 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);
/** Save audio*/
static void saveBuffer(void *user_data);
// Restart the pipeline /** Restart the pipeline */
static gboolean reconnectURIStream(void *data); static gboolean reconnectURIStream(void* data);
public: public:
StreamRecorder(const char* source, int time); StreamRecorder(const char* source, int time);
}; };
//------------------------------------------------------------------------------ //------------------------------------------------------------------------------
......
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