add format

StreamRecorder.cpp

@@ -139,7 +139,7 @@ int StreamRecorder::createMainPipeline() {
     gst_bin_add_many (GST_BIN (mainPipeline), streamSrc, audioConvert, filterCaps, queue0, filter,queue1, fakeSink, NULL);
     if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, fakeSink,  NULL)){
         //gst_bin_add_many (GST_BIN (mainPipeline), streamSrc, audioConvert, filterCaps, queue0, filter, queue1, audioSink, NULL);
-    //if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, audioSink,  NULL)){
+        // if(!gst_element_link_many(audioConvert, filterCaps, queue0, filter, queue1, audioSink,  NULL)){
         
         cerr << "mainPipeline: Failed to link elements in the pipeline" << endl;
 		exit(0);
@@ -171,7 +171,6 @@ 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){
@@ -256,8 +255,8 @@ void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *
     
     GstPad *sinkpad;
     
-    // gst_pad_get_caps is for gst v0.1
-	//GstCaps *new_pad_caps = gst_pad_get_caps (pad);
+    //gst_pad_get_caps is for gst v0.1
+    // GstCaps *new_pad_caps = gst_pad_get_caps (pad);
     
     GstCaps *new_pad_caps = gst_pad_query_caps(pad, NULL);
     GstStructure *new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
@@ -270,7 +269,6 @@ void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *
         sinkpad = gst_element_get_static_pad (nextElement, "sink");
         
         if (GST_PAD_LINK_FAILED (gst_pad_link (pad, sinkpad))) {
-   
             cerr << "Type is "<< new_pad_type <<" but link failed." << endl;
             exit(0);
         }
@@ -286,8 +284,8 @@ void StreamRecorder::srcNewPad_callback(GstElement *element, GstPad *pad, void *
 *	@return unimplemented
 */
 
-int StreamRecorder::compressBuffer()
-{
+int StreamRecorder::compressBuffer() {
+    
     long int currentTime = time(NULL);
     stringstream ss;
     string fileNameStr, currentTimeStr;
@@ -305,11 +303,11 @@ int StreamRecorder::compressBuffer()
     FLAC__StreamEncoderInitStatus init_status;
     
     /* allocate the encoder */
-	if((encoder = FLAC__stream_encoder_new()) == NULL)
-	{
+    if((encoder = FLAC__stream_encoder_new()) == NULL) {
         fprintf(stderr, "ERROR: allocating encoder\n");
         return 1;
 	}
+    
     ok &= FLAC__stream_encoder_set_verify(encoder, true);
     ok &= FLAC__stream_encoder_set_compression_level(encoder, 5);
     ok &= FLAC__stream_encoder_set_channels(encoder, 1);
@@ -318,8 +316,7 @@ int StreamRecorder::compressBuffer()
     ok &= FLAC__stream_encoder_set_total_samples_estimate(encoder, nBytes/STREAMRECORDER_BYTESPERSAMPLE);
     
     /* now add some metadata; we'll add some tags and a padding block */
-	if(ok)
-	{
+    if(ok) {
         if(
             (metadata[0] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_VORBIS_COMMENT)) == NULL ||
             (metadata[1] = FLAC__metadata_object_new(FLAC__METADATA_TYPE_PADDING)) == NULL ||
@@ -357,19 +354,16 @@ int StreamRecorder::compressBuffer()
 	int* pcm = new int[readsize*channels];
 	/* read blocks of samples from WAVE file and feed to encoder */
 
-	if(ok)
-	{
+	if(ok) {
         int left = nBytes;
         
-		while(left&&ok)
-		{
+        while(left&&ok) {
             
             int need = (left>readsize? readsize : left);
             memcpy(buffer, audioBufferTmp, need);//*sizeof(short));
             audioBufferTmp+=need;//*sizeof(short);
-			/* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
-			for(int i=0; i < need*channels; i++)
-			{
+            // /* convert the packed little-endian 16-bit PCM samples from WAVE into an interleaved FLAC__int32 buffer for libFLAC */
+            for(int i=0; i < need*channels; i++) {
                 /* inefficient but simple and works on big- or little-endian machines */
                 pcm[i] = (FLAC__int32)(((FLAC__int16)(FLAC__int8)buffer[2*i+1] << 8) | (FLAC__int16)buffer[2*i]);
 			}
@@ -378,8 +372,7 @@ int StreamRecorder::compressBuffer()
 			left -= need;
 		}
 	}
-	else
-	{
+	else {
         cout << "-ERROR-" << endl;
 	}
     
@@ -419,17 +412,14 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) {
     
     audioBufferPosition+=bytesRead;
     nBytes+=bytesRead;//READSIZE;
-    
     cout << "Bytes readed " << nBytes << endl;
     cout << "Buffer size " << bufferSize << endl;
     
     if(nBytes >= bufferSize) {
-    
         compressBuffer();
         audioBufferPosition=audioBuffer;
         memset (audioBuffer, 0, bufferSize);
         nBytes=0;
-  
     }
     
     return nBytes;
@@ -446,8 +436,8 @@ int StreamRecorder::addToBuffer(unsigned char* data, int length) {
 *	@return unimplemented
 */
 
-int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffer, void* user_data)
-{
+int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffer, void* user_data) {
+    
     cout << "LLEGO INFO" << endl;
     GstMapInfo info;
     if(!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
@@ -455,7 +445,7 @@ int StreamRecorder::filter_handoff_callback(GstElement* filter, GstBuffer* buffe
     }
     
     //GST_BUFFER_DATA is for gst v0.1
-	//((StreamRecorder*)user_data)->addToBuffer((unsigned char*)GST_BUFFER_DATA (buffer));
+    // ((StreamRecorder*)user_data)->addToBuffer((unsigned char*)GST_BUFFER_DATA (buffer));
     
     // user data is the class
     ((StreamRecorder*)user_data)->addToBuffer((unsigned char*)info.data, info.size);