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GStreamer_audioRecorder
GStreamer_audioRecorder
Commit a187cafe authored by d.basulto's avatar d.basulto
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change the way that the bytes to add is calculated

parent 262b856b
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Showing with 142 additions and 185 deletions
StreamRecorder.cpp
View file @ a187cafe
......@@ -17,6 +17,8 @@
#include <glib.h>
#include <unistd.h>
#include <cstring>
#include <cstdint>
#include <cstdio>
#include "FLAC/metadata.h"
#include "FLAC/stream_encoder.h"
......@@ -43,6 +45,8 @@ StreamRecorder::StreamRecorder(const char* source, int time)
bufferSize=nFrames*READSIZE*STREAMRECORDER_BYTESPERSAMPLE;
audioBuffer = new unsigned char[bufferSize];
bytesPerSecond= bufferSize/audioFileDuration;
// New buffer
memset(audioBuffer, 0, bufferSize);
audioBufferPosition=audioBuffer;
......@@ -64,7 +68,7 @@ int StreamRecorder::connect()
{
disconnect();
cout << "connecting to " << pluginUri << endl;
cout << "connecting to " << pluginUri << "..." << endl;
gst_element_set_state (mainPipeline, GST_STATE_NULL);
gst_element_set_state (mainPipeline, GST_STATE_READY);
......@@ -151,7 +155,7 @@ int StreamRecorder::createMainPipeline()
*/
gboolean StreamRecorder::reconnectURIStream(void* instance)
{
cout << "\nTrying to reconnect with the stream..." << endl << endl;
cout << "Trying to reconnect with the stream..." << endl << endl;
((StreamRecorder*)instance)->connect();
return FALSE;
......@@ -170,160 +174,70 @@ gboolean StreamRecorder::reconnectURIStream(void* instance)
int StreamRecorder::bus_callback (GstBus *bus, GstMessage *message, void *user_data)
{
//printf("StreamRecorder got %s message\n", GST_MESSAGE_TYPE_NAME (message));
if(GST_MESSAGE_TYPE (message) == GST_MESSAGE_EOS)
{
((StreamRecorder*)user_data)->isConnectionLost = true;
}
switch (GST_MESSAGE_TYPE (message))
{
case GST_MESSAGE_EOS:
cout << "** End of stream **\n" << endl;
cout << "** End of stream **" << endl << endl;
if(((StreamRecorder*)user_data)->isConnectionLost) //Enter only if the connection is lost
{
// ----------------------------------------------------------------------------------------
long int actualRecordTime;
long int currentTime = time(NULL);
long int addTime;
long int bytesToAdd;
//cout << "Audio start time (timestamp): " << ((StreamRecorder*)user_data)->timestamp << endl;
//cout << "Actual time (timestamp): " << currentTime << endl;
actualRecordTime = currentTime-((StreamRecorder*)user_data)->timestamp;
cout << "Record time: " << actualRecordTime << endl;
// ----------------------------------------------------------------------------------------
/** The calculated record time is greater? */
if(actualRecordTime >= ((StreamRecorder*)user_data)->audioFileDuration)
{
cout << "Bytes readed: " << ((StreamRecorder*)user_data)->nBytes << endl;
saveBuffer(user_data);
}
else
{
/** Moves the pointer to the position corresponding to the difference of the timestamps */
addTime = currentTime - ((StreamRecorder *) user_data)->oldTmpTimestamp;
cout << "Time to add: " << addTime << endl;
bytesToAdd = addTime*READSIZE*STREAMRECORDER_BYTESPERSAMPLE*10;
cout << "-------------------------------------" << endl;
cout << "Actual Bytes number: " << "(" << ((StreamRecorder*)user_data)->nBytes << ") + ";
cout << "Bytes t/add: " << "(" << bytesToAdd << ")" << endl;
/** Update the pointer and the bytes number */
((StreamRecorder*)user_data)->nBytes+=bytesToAdd;
((StreamRecorder*)user_data)->audioBufferPosition+=bytesToAdd;
cout << "Result : " << ((StreamRecorder*)user_data)->nBytes << endl;
cout << "Buffer size: " << ((StreamRecorder*)user_data)->bufferSize << endl;
cout << "-------------------------------------" << endl;
}
((StreamRecorder*)user_data)->isConnectionLost = false;
/** Try reconnect with the radio stream*/
g_timeout_add(RECONNECTION_DELAY*1000, reconnectURIStream, user_data);
}
((StreamRecorder*)user_data)->isDisconnected = true;
break;
case GST_MESSAGE_ERROR:
gchar *debug;
GError *error;
gchar *debug;
gst_message_parse_error (message, &error, &debug);
g_free (debug);
strcpy(((StreamRecorder*)user_data)->errorMessage,error->message);
/** The message doesn't contains null character*/
((StreamRecorder*)user_data)->errorMessage[ERROR_MSG_SIZE-1] = '\0';
//cerr << "Error: "<< ((StreamRecorder*)user_data)->errorMessage << endl;
g_error_free (error);
cerr << "Error - code: (" << error->code << ") - Message:" << error->message << endl;
if(strcmp(((StreamRecorder*)user_data)->errorMessage, "Stream doesn't contain enough data.") == 0)
switch (error->code)
{
/** Not audio stream received */
if(((StreamRecorder*)user_data)->timestamp == 0)
{
g_timeout_add(RECONNECTION_DELAY*1000, reconnectURIStream, user_data);
}
else
{
// ----------------------------------------------------------------------------------------
long int actualRecordTime;
long int currentTime = time(NULL);
//long int addTime;
long int bytesToAdd;
actualRecordTime = currentTime - ((StreamRecorder *)user_data)->timestamp;
cout << "Record time: " << actualRecordTime << endl;
// ----------------------------------------------------------------------------------------
/** Record time is greater than th required */
if (actualRecordTime >= ((StreamRecorder *) user_data)->audioFileDuration)
{
cout << "Bytes readed: " << ((StreamRecorder *) user_data)->nBytes << endl;
saveBuffer(user_data);
}
else
{
/** Moves the pointer to the position corresponding to the difference of the timestamps */
//addTime = currentTime - ((StreamRecorder *) user_data)->oldTmpTimestamp;
cout << "Time to add: " << RECONNECTION_DELAY << endl;
bytesToAdd = RECONNECTION_DELAY*READSIZE*STREAMRECORDER_BYTESPERSAMPLE*10;
cout << "-------------------------------------" << endl;
cout << "Actual Bytes number: " << "(" << ((StreamRecorder *) user_data)->nBytes << ") + ";
cout << "Bytes t/add: " << "(" << bytesToAdd << ")" << endl;
/** Update the pointer and the bytes number */
((StreamRecorder *) user_data)->nBytes += bytesToAdd;
((StreamRecorder *) user_data)->audioBufferPosition += bytesToAdd;
cout << "Result : " << ((StreamRecorder *) user_data)->nBytes << endl;
cout << "Buffer size: " << ((StreamRecorder *) user_data)->bufferSize << endl;
cout << "-------------------------------------" << endl;
}
((StreamRecorder*)user_data)->isConnectionLost = false;
/** Try reconnect with the radio stream*/
g_timeout_add(RECONNECTION_DELAY*1000, reconnectURIStream, user_data);
}
case STREAM_ENOUGH_DATA_ERROR:
/** Last message in errors secuence*/
((StreamRecorder*)user_data)->isDisconnected = true;
break;
default:
return TRUE;
}
g_error_free (error);
break;
default:
break;
return TRUE;
}
if(((StreamRecorder*)user_data)->isDisconnected)
{
/** Try reconnect with the radio stream*/
cout << "\nReconnection started..." << endl;
g_timeout_add(RECONNECTION_DELAY*1000, reconnectURIStream, user_data);
}
return TRUE;
}
//------------------------------------------------------------------------------
void StreamRecorder::saveBuffer(void *user_data)
{
int missingBytes = ((StreamRecorder*)user_data)->bufferSize - ((StreamRecorder*)user_data)->nBytes;
//((StreamRecorder*)user_data)->audioBufferPosition+=missingBytes;
((StreamRecorder*)user_data)->nBytes+=missingBytes;
((StreamRecorder*)user_data)->compressBuffer();
((StreamRecorder*)user_data)->audioBufferPosition=((StreamRecorder*)user_data)->audioBuffer;
memset (((StreamRecorder*)user_data)->audioBuffer, 0, ((StreamRecorder*)user_data)->bufferSize);
((StreamRecorder*)user_data)->nBytes=0;
/**
* Do all the buffer saving and restart operations
*/
void StreamRecorder::saveBuffer()
{
compressBuffer();
audioBufferPosition=audioBuffer;
memset (audioBuffer, 0, bufferSize);
nBytes=0;
}
//------------------------------------------------------------------------------
/**
* CallBack to link the pads created by uridecodebin
......@@ -361,6 +275,7 @@ void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *
cout <<"Link succeeded " << new_pad_type << endl;
}
}
//------------------------------------------------------------------------------
/**
......@@ -370,15 +285,50 @@ void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *
* @param user_data this
* @return unimplemented
*/
int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffer, void* user_data)
{
GstMapInfo info;
if(!gst_buffer_map (buffer, &info, GST_MAP_READ))
cout << "ERROR: MAPPING IS NOT VALID" << endl;
long int currentTime;
long int actualRecordTime;
//GST_BUFFER_DATA is for gst v0.1
// ((StreamRecorder*)user_data)->addToBuffer((unsigned char*)GST_BUFFER_DATA (buffer));
if(((StreamRecorder*)user_data)->isDisconnected){
currentTime = time(NULL);
actualRecordTime = currentTime - ((StreamRecorder*)user_data)->timestamp;
if(actualRecordTime >= ((StreamRecorder*)user_data)->audioFileDuration)
{
/** Complete the buffer with missing bytes*/
((StreamRecorder*)user_data)->nBytes = ((StreamRecorder*)user_data)->bufferSize;
((StreamRecorder*)user_data)->saveBuffer();
cout << "The buffer has been saved..." << endl;
}
else
{
long int bytesToAdd = actualRecordTime*((StreamRecorder*)user_data)->bytesPerSecond;
unsigned char *newPosition;
printf("Bytes to add %d\n", bytesToAdd);
newPosition = ((StreamRecorder*)user_data)->audioBuffer;
newPosition += bytesToAdd;
((StreamRecorder*)user_data)->audioBufferPosition = newPosition;
((StreamRecorder*)user_data)->nBytes = bytesToAdd;
cout << "The pointer has been updated..." << endl;
}
((StreamRecorder*)user_data)->isDisconnected = false;
}
// user data is the class
((StreamRecorder*)user_data)->addToBuffer((unsigned char*)info.data, info.size);
gst_buffer_unmap (buffer, &info);
......@@ -395,8 +345,6 @@ int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffe
int StreamRecorder::addToBuffer(unsigned char* data, int length) {
int bytesRead = length;// READSIZE*STREAMRECORDER_BYTESPERSAMPLE;
long int currentTime;
long int actualRecordTime;
bool isNewAudioFile;
/** Useful for obtain the filename*/
......@@ -404,13 +352,13 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) {
cout << "Data size: " << bytesRead << endl;
//if(nBytes < bufferSize)
//{
/** Update pointer*/
/** add info to buffer*/
printf("Audio buffer position %d\n", audioBufferPosition);
memcpy((char*)audioBufferPosition, (char*)data, bytesRead);
/** Update pointer*/
nBytes+=bytesRead;
audioBufferPosition+=bytesRead;
//}
if(isNewAudioFile)
{
......@@ -419,14 +367,14 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) {
/** filename */
timestamp = time(NULL);
cout << "Audio filename (timestamp): " << timestamp << endl;
printf("Audio buffer %d\n", audioBuffer);
}
else
{
long int currentTime;
long int actualRecordTime;
currentTime = time(NULL);
oldTmpTimestamp = newTmpTimestamp;
newTmpTimestamp = currentTime;
actualRecordTime = currentTime-timestamp;
cout << "Record time: " << actualRecordTime << endl;
......@@ -436,10 +384,12 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) {
/** If the buffer is full, save it in flac file */
if(nBytes >= bufferSize)
{
compressBuffer();
audioBufferPosition=audioBuffer;
memset (audioBuffer, 0, bufferSize);
nBytes=0;
cout << "ENTROOOO" << endl;
printf("Audio buffer %d\n", audioBuffer );
printf("Audio buffer position %d\n", audioBufferPosition);
cout << "Bytes readed " << nBytes << endl;
cout << "Buffer size " << bufferSize << endl;
saveBuffer();
}
}
......
StreamRecorder.h
View file @ a187cafe
......@@ -28,6 +28,8 @@
#define DST_URI_SIZE 80
#define RECONNECTION_DELAY 1
#define STREAM_ENOUGH_DATA_ERROR 4
//------------------------------------------------------------------------------
/** Class declaration */
......@@ -36,60 +38,65 @@ class StreamRecorder
{
private:
unsigned char* audioBuffer;
unsigned char* audioBufferPosition;
unsigned char* audioBuffer;
unsigned char* audioBufferPosition;
char errorMessage[ERROR_MSG_SIZE];
char pluginUri[DST_URI_SIZE];
char errorMessage[ERROR_MSG_SIZE];
char pluginUri[DST_URI_SIZE];
unsigned int nBytes;
unsigned int bufferSize;
unsigned int nBytes;
unsigned int bufferSize;
int recordTime;
int audioFileDuration;
/** Audio filename */
long int timestamp = 0;
long int oldTmpTimestamp = 0;
long int newTmpTimestamp = 0;
int recordTime;
int audioFileDuration;
bool isConnectionLost;
//char* sourceName;
//GstElement* audioResample;
//GstElement* tempBin;
//GstElement* audioSink;
GstElement* streamSrc;
GstElement* audioConvert;
GstElement* filterCaps;
GstElement* queue0;
GstElement* queue1;
GstElement* filter;
GstElement* fakeSink;
GstElement* mainPipeline;
/** Audio filename */
long int timestamp = 0;
long int oldTmpTimestamp = 0;
long int newTmpTimestamp = 0;
long int bytesPerSecond = 0;
bool isDisconnected = false;
bool isValidDisconnectedEvent = false;
int createMainPipeline();
int connect();
int disconnect();
//char* sourceName;
//GstElement* audioResample;
//GstElement* tempBin;
//GstElement* audioSink;
GstElement* streamSrc;
GstElement* audioConvert;
GstElement* filterCaps;
GstElement* queue0;
GstElement* queue1;
GstElement* filter;
GstElement* fakeSink;
GstElement* mainPipeline;
int createMainPipeline();
int connect();
int disconnect();
/** add data to buffer */
int addToBuffer(unsigned char* data, int length);
int compressBuffer();
/** 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);
/** 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);
/** Save audio*/
static void saveBuffer(void *user_data);
/** Save audio*/
void saveBuffer();
/** Restart the pipeline */
static gboolean reconnectURIStream(void* data);
/** Restart the pipeline */
static gboolean reconnectURIStream(void* data);
public:
public:
StreamRecorder(const char* source, int time);
StreamRecorder(const char* source, int time);
};
//------------------------------------------------------------------------------
#endif
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