value output from audio analysis

1 files changed, 135 insertions, 0 deletions

diff --git a/rotord/src/nodes_audio_analysis.cpp b/rotord/src/nodes_audio_analysis.cpp
new file mode 100644
index 0000000..925858f
--- /dev/null
+++ b/rotord/src/nodes_audio_analysis.cpp
@@ -0,0 +1,135 @@
+#include "nodes_audio_analysis.h"
+
+namespace Rotor{
+	bool Audio_thumbnailer::init(int _channels,int _bits,int _samples,int _rate) {
+		//base_audio_processor::init(_channels,_bits,_samples);
+		channels=_channels;
+		bits=_bits;
+		samples=_samples;
+		samples_per_column=samples/width;
+		column=0; //point thumbnail bitmap
+		out_sample=0; //sample in whole track
+		offset=0x1<<(bits-1); //signed audio
+		scale=1.0f/offset;
+		sample=0;
+		samples=0;
+		accum=0.0;
+		return true;
+	}
+	int Audio_thumbnailer::process_frame(uint8_t *_data,int samples_in_frame){
+		//begin by processing remaining samples
+		//samples per column could be larger than a frame! (probably is)
+		//but all we are doing is averaging
+		int bytes=(bits>>3);
+		int stride=channels*bytes;
+		int in_sample=0;
+		while (in_sample<samples_in_frame&&column<width) {
+			//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[(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...
+					double val=((double)((int16_t)this_val))*scale;
+					accum+=val*val;
+					samples++;
+				}
+				in_sample++;
+				sample++;
+				out_sample++;
+			}
+			if (sample==samples_per_column) { //finished a column
+				//get root-mean
+				double mean=pow(accum/samples,0.5);
+				//if (column==0) {
+				//	cerr << "first column total: "<< accum << " in " << samples << " samples, average " << (accum/samples)<<endl;
+				//}
+				int colheight=height*mean*0.5;
+				int hh=height>>1;
+				for (int i=0;i<height;i++) {
+					data[i*width+column]=abs(i-hh)<colheight?0xff:0x00;
+				}
+				vectordata[column]=mean;
+				column++;
+				sample=0;
+				samples=0;
+				accum=0.0;
+			}
+		}
+		return out_sample;
+	}
+	string Audio_thumbnailer::print(){
+		//base64 encode the image data output it
+
+		stringstream output;
+		Poco::Base64Encoder *enc=new Poco::Base64Encoder(output);
+
+		enc->write((char*)data,width*height);
+		//tring output;
+		/*
+		for (int j=0;j<height;j++) {
+			for (int i=0;i<width;i++) {
+				output+=data[j*width+i]<0x7f?"0":"1";
+			}
+			output +="\n";
+		}
+		*/
+		enc->close();
+		delete enc;
+		return output.str();
+	}
+	void Audio_thumbnailer::print_vector(xmlIO XML){
+		string vdata;
+		for (int i=0;i<width;i++){
+			if (i>0) vdata+=",";
+			vdata+=ofToString(vectordata[i]);
+		}
+		XML.addValue("data",vdata);
+	}
+	bool Audio_analysis::init(int _channels,int _bits,int _samples, int _rate) {
+		//need these to make sense of data
+		channels=_channels;
+		bits=_bits;
+		samples=_samples;
+
+		return analyser.init(soname,id,_channels,_bits,_samples,_rate,outputNo,params);
+
+
+		//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
+
+	}
+	int Audio_analysis::process_frame(uint8_t *data,int samples_in_frame) {
+		analyser.process_frame(data,samples_in_frame);
+		return 1;
+	}
+	void Audio_analysis::cleanup() {
+		analyser.cleanup();
+		//print_features();
+	}
+	string Audio_analysis::get_features(){
+		string data;
+		for (auto i: analyser.features) {
+			data=data+" ["+ofToString(i.second.number)+":"+ofToString(i.first);
+			if (i.second.values.size()) {
+				data+=" (";
+				bool first=true;
+				for (auto j: i.second.values) {	
+					if (first){
+						first=false;
+					}
+					else data+=",";
+					data=data+ofToString(j);
+				}
+				data+=") ";
+			}	
+			data+="]";
+		}
+		return data;
+	}
+}
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