thumbnailing

1 files changed, 12 insertions, 14 deletions

diff --git a/rotord/rotor.cpp b/rotord/rotor.cpp
index 73577f4..2cd894f 100644
--- a/rotord/rotor.cpp
+++ b/rotord/rotor.cpp
@@ -259,7 +259,7 @@ Command_response Render_context::session_command(const std::vector<std::string>&
 //great to use c++11 features
 
 bool Render_context::load_audio(const string &filename,vector<Base_audio_processor*> processors){
-	
+
 	av_register_all();
 
 	AVFrame* frame = avcodec_alloc_frame();
@@ -363,22 +363,22 @@ bool Render_context::load_audio(const string &filename,vector<Base_audio_process
 				// the audio data, so I won't add any junk here that might confuse you. Typically, if I want to find
 				// documentation on an FFmpeg structure or function, I just type "<name> doxygen" into google (like
 				// "AVFrame doxygen" for AVFrame's docs)
-				
+
 				//av_get_channel_layout_string (char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
-				
-				
+
+
 				//now we can pass the data to the processor(s)
 				for (auto p: processors) {
 					sample_processed=p->process_frame(frame->data[0],frame->nb_samples);
 				}
-				
+
 				mutex.lock();
 				progress=((double)sample_processed)/samples;
 				mutex.unlock();
 			}
 		}
 		// You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
-		av_free_packet(&packet);  
+		av_free_packet(&packet);
 	}
 
 	// Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
@@ -399,14 +399,14 @@ bool Render_context::load_audio(const string &filename,vector<Base_audio_process
 			mutex.unlock();
 		}
 	}
-	
+
 	cerr << "finished processing: "<<sample_processed << " samples of  "<<samples<<", "<<((double)sample_processed*100)/samples<<"%"<<  std::endl;
-					
+
 	// Clean up!
 	for (auto p: processors) {
 		p->cleanup();
 	}
-				
+
 	av_free(frame);
 	avcodec_close(codecContext);
 	av_close_input_file(formatContext);
@@ -471,7 +471,6 @@ bool Graph::load(string &filename){
 	}
 	else return false;
 }
-
 Node_factory::Node_factory(){
 	//for now, statically load prototype map in constructor
 	add_type("audio_analysis",new Audio_analysis());
@@ -502,14 +501,13 @@ int Audio_thumbnailer::process_frame(uint8_t *_data,int samples_in_frame){
 	int stride=channels*bytes;
 	int in_sample=0;
 	while (in_sample<samples_in_frame&&column<width) {
-		//start a new column
-		
+		//continue the column
 		while (sample<samples_per_column&&in_sample<samples_in_frame) {
 			//accumulate samples for this column until we run out of samples
 			for (int i=0;i<channels;i++) {
 				unsigned int this_val=0;
 				for (int j=0;j<bytes;j++) {
-					this_val+=_data[(sample*stride)+(i*bytes)+j]<<(j*8);
+					this_val+=_data[(in_sample*stride)+(i*bytes)+j]<<(j*8);
 				}
 				//convert from integer data format - i.e s16p - to audio signal in -1..1 range
 				//presume 16 bits for now...
@@ -557,7 +555,7 @@ bool Audio_analysis::init(int _channels,int _bits,int _samples, int _rate) {
 	samples=_samples;
 
 	return analyser.init(soname,id,_channels,_bits,_samples,_rate);
-	
+
 	//attempt to load vamp plugin and prepare to receive frames of data
 	//should the audio analysis contain a vamphost or should it inherit?
 	//maybe neater to contain it in terms of headers etc