From 1e26d66c3a7be31a9dd214b587488dfe3ed69b4b Mon Sep 17 00:00:00 2001
From: Tim Redfern <tim@herge.(none)>
Date: Tue, 23 Apr 2013 14:01:06 +0100
Subject: writing audio but sped up for some reason

---
 rotord/libavaudioloader.cpp | 45 ++++++++++++++++++++++++++++++++++++++-------
 1 file changed, 38 insertions(+), 7 deletions(-)

(limited to 'rotord/libavaudioloader.cpp')

diff --git a/rotord/libavaudioloader.cpp b/rotord/libavaudioloader.cpp
index 133fcd4..b842a42 100644
--- a/rotord/libavaudioloader.cpp
+++ b/rotord/libavaudioloader.cpp
@@ -100,6 +100,30 @@ bool libav::Audioloader::setup(const std::string &filename){
 		return frame;
 
 	}
+	AVPacket* libav::Audioloader::get_packet() {
+
+		if (!ready) return nullptr;
+
+		int ret=av_read_frame(formatContext, &packet);
+        if (ret<0) {
+            std::cerr << "finished with code "<<ret <<(ret==AVERROR_EOF?" ,EOF":"")<<std::endl;
+            ready=false;
+            return nullptr;
+        }
+        //if (packet.stream_index == audioStream->index)
+        //{
+            //int bytes =
+        //   avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet);
+
+            // Some frames rely on multiple packets, so we have to make sure the frame is finished before
+            // we can use it
+        //}
+        // You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
+        //av_free_packet(&packet);?????
+		//}
+		return &packet;
+
+	}
 
 /*
 		// Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
@@ -130,8 +154,8 @@ uint16_t* libav::Audioloader::get_samples(int num){ //presumes 16bpc here
 	//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];
+			for (int j=0;j<av_frame_get_channels(frame);j++) {
+				buffer[(i*av_frame_get_channels(frame))+j]=buffer[((sample_start+i)*av_frame_get_channels(frame))+j];
 			}
 		}
 		sample_start=sample_end-sample_start;
@@ -139,18 +163,25 @@ uint16_t* libav::Audioloader::get_samples(int num){ //presumes 16bpc here
 	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 (((sample_end+std::max(num,frame->nb_samples))*av_frame_get_channels(frame))>buffer.size()){
+			int m=buffer.size();
+			int s=((sample_end+std::max(num,frame->nb_samples))*av_frame_get_channels(frame));
+			buffer.reserve(s);
+			std::cerr << "audioloader reserved buffer to " << s << std::endl;
+			for (int i=m;i<s;i++) buffer.push_back(0);
 		}
 		if (!frame) {
-			for (int i=0;i<num*frame->channels;i++){
+			for (int i=0;i<num*av_frame_get_channels(frame);i++){
 				buffer[sample_end+i]=0;
 			}
 
 		}
+		//std::cerr<<"filling buffer to "<<((sample_end+frame->nb_samples)*frame->channels)<<std::endl;
 		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];
+			for (int j=0;j<av_frame_get_channels(frame);j++) {
+				uint16_t s=((uint16_t*) frame->buf[j]->data)[i];
+				//uint16_t s=frame->buf[j]->data[i*2]+((frame->buf[j]->data)[i*2+1]<<8);
+				buffer[((sample_end+i)*frame->channels)+j]= s;
 			}
 		}
 		sample_end+=frame->nb_samples;
-- 
cgit v1.2.3