audio output corruption

1 files changed, 51 insertions, 10 deletions

diff --git a/rotord/libavaudioloader.cpp b/rotord/libavaudioloader.cpp
index 8a2d81a..133fcd4 100644
--- a/rotord/libavaudioloader.cpp
+++ b/rotord/libavaudioloader.cpp
@@ -1,6 +1,6 @@
 #include "libavaudioloader.h"
 
-bool libav::audioloader::setup(const std::string &filename){
+bool libav::Audioloader::setup(const std::string &filename){
 
 		av_register_all();
 
@@ -65,16 +65,16 @@ bool libav::audioloader::setup(const std::string &filename){
 
 		av_dump_format(formatContext, 0, 0, false); //avformat.h line 1256
 		int samples = ((formatContext->duration + 5000)*codecContext->sample_rate)/AV_TIME_BASE;
-		
+
 		std::cout << "This stream has " << codecContext->channels << " channels, a sample rate of " << codecContext->sample_rate << "Hz and "<<samples <<" samples" << std::endl;
 		std::cout << "The data is in format " <<codecContext->sample_fmt<< " (aka "<< av_get_sample_fmt_name(codecContext->sample_fmt) << ") "<<std::endl;
 
 		av_init_packet(&packet);
-		sample_processed=0;
+		//sample_processed=0;
 		ready=true;
 		return true;
 	}
-	AVFrame* libav::audioloader::get_frame() {
+	AVFrame* libav::Audioloader::get_frame() {
 
 		if (!ready) return nullptr;
 
@@ -124,10 +124,51 @@ bool libav::audioloader::setup(const std::string &filename){
 		cerr << "finished processing: "<<sample_processed << " samples of  "<<samples<<", "<<((double)sample_processed*100)/samples<<"%"<<  std::endl;
 */
 
-	bool libav::audioloader::close() {
-		av_free(frame);
-		avcodec_close(codecContext);
-		avformat_close_input(&formatContext);
+uint16_t* libav::Audioloader::get_samples(int num){ //presumes 16bpc here
+	//std::cerr << "request "<<num<<" samples: "<<(ready?"ready":"not ready")<<std::endl;
+	if(!ready) return nullptr;
+	//shuffle down samples
+	if (sample_start>0){
+		for (int i=0;i<sample_end-sample_start;i++){
+			for (int j=0;j<frame->channels;j++) {
+				buffer[(i*frame->channels)+j]=buffer[((sample_start+i)*frame->channels)+j];
+			}
+		}
+		sample_start=sample_end-sample_start;
+	}
+	sample_end=sample_start;
+	while (sample_end<num) {
+		frame=get_frame();
+		if (((sample_end+frame->nb_samples)*frame->channels)>buffer.size()){
+			buffer.reserve((sample_end+frame->nb_samples)*frame->channels);
+		}
+		if (!frame) {
+			for (int i=0;i<num*frame->channels;i++){
+				buffer[sample_end+i]=0;
+			}
 
-		return true;
-	}
\ No newline at end of file
+		}
+		for (int i=0;i<frame->nb_samples;i++) {
+			for (int j=0;j<frame->channels;j++) {
+				buffer[((sample_end+i)*frame->channels)+j]= ((uint16_t*) frame->buf[0])[(i*frame->channels)+j];
+			}
+		}
+		sample_end+=frame->nb_samples;
+		//avcodec_free_frame(&frame);
+	}
+	if (sample_end>num) {
+		sample_start=num;
+	}
+	else {
+		sample_start=0;
+	}
+	return (uint16_t*)(&buffer[0]);
+}
+
+bool libav::Audioloader::close() {
+	av_free(frame);
+	avcodec_close(codecContext);
+	avformat_close_input(&formatContext);
+
+	return true;
+}