From f4c9ea89fa6d642813dc06387e956ef062ad9de1 Mon Sep 17 00:00:00 2001
From: Tim Redfern <tim@herge.(none)>
Date: Tue, 2 Apr 2013 17:14:27 +0100
Subject: baffled by premature EOF

---
 rotord/rotor.cpp | 60 +++++++++++++++++++++++++++++++++++++++++++++++++++-----
 1 file changed, 55 insertions(+), 5 deletions(-)

(limited to 'rotord/rotor.cpp')

diff --git a/rotord/rotor.cpp b/rotord/rotor.cpp
index 2ab5734..53ba33b 100644
--- a/rotord/rotor.cpp
+++ b/rotord/rotor.cpp
@@ -431,9 +431,31 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
 		*/
 
 
-		http://www.gamedev.net/topic/624876-how-to-read-an-audio-file-with-ffmpeg-in-c/
+		//http://www.gamedev.net/topic/624876-how-to-read-an-audio-file-with-ffmpeg-in-c/
 		// Initialize FFmpeg
+		
+		//this seems to only read a certain percentage of the sound file? - different for wav and mp3
+		//peculiar, because the data comes in as frames and we just read them until all are done
+		//is it possible that thhe problem lies with the drawing process only?
+		//wav sees 6/10 waves and mp3 sees 3.5/10
+		//could this be an aliasing effect i.e. from reading the sample format wrong?
+		//I think probably not?
+		
+		//every format combination seems to miss a different number of samples
+		//I don't understand how different formats would make the samples seem to truncate?
+		//unless its an aliasing type thing?
+		
+		//wave.mp3 reads 16.409% of 160913 
+		//wave.wav reads 29.706% of 882044
+		
+		
+		
 	av_register_all();
+	
+	//??
+	//avcodec_init();
+	avcodec_register_all();
+	//??
 
 	AVFrame* frame = avcodec_alloc_frame();
 	if (!frame)
@@ -497,6 +519,12 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
 		std::cout << "Couldn't open the context with the decoder" << std::endl;
 		return false;
 	}
+	
+	//
+	// why is the file truncated???
+	//if(codecContext->codec->capabilities & CODEC_CAP_TRUNCATED) codecContext->codec->capabilities|=CODEC_FLAG_TRUNCATED;
+	//
+	//
 
 	av_dump_format(formatContext, 0, 0, false); //avformat.h line 1256
 	int samples = ((formatContext->duration + 5000)*codecContext->sample_rate)/AV_TIME_BASE;
@@ -514,10 +542,19 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
 	AVPacket packet;
 	av_init_packet(&packet);
 	int sample_processed=0;
+	
+	bool diag=true;
 
 	// Read the packets in a loop
-	while (av_read_frame(formatContext, &packet) == 0)
+	while (true)
+	//while(sample_processed<samples)
 	{
+		int ret=av_read_frame(formatContext, &packet);
+		if (ret<0) {
+			cerr << "finished with code "<<ret <<(ret==AVERROR_EOF?" ,EOF":"")<<endl;
+			break;
+		}
+		//av_read_frame(formatContext, &packet); //hangs once the packets have been read
 		if (packet.stream_index == audioStream->index)
 		{
 			// Try to decode the packet into a frame
@@ -535,12 +572,20 @@ 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)
+				
+				if (diag) {
+					cerr << "first frame: "<<bytes << ", "<<frame->nb_samples<<" samples in "<<av_get_sample_fmt_name(frame->format)<<" format with channel layout "<<frame->channel_layout<<  std::endl;
+					diag=false;
+				}
 
 				//std::cout << "Got a frame: bytes " << bytes << ", "<<frame->nb_samples<<" samples"<<std::endl;
 				//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();
@@ -568,6 +613,9 @@ bool Render_context::load_audio(const string &filename,vector<base_audio_process
 			mutex.unlock();
 		}
 	}
+	
+	cerr << "finished processed: "<<sample_processed << " samples of  "<<samples<<" , "<<((double)sample_processed*100)/samples<<"%"<<  std::endl;
+					
 
 	// Clean up!
 	av_free(frame);
@@ -638,6 +686,8 @@ void audio_thumbnailer::init(int _channels,int _bits,int _samples) {
 	channels=_channels;
 	bits=_bits;
 	samples=_samples;
+	
+	cerr << "init audio thumbnailer with "<<channels<<" channels, "<<samples<<" samples of "<< bits<<" bits."<<endl;
 
 
 	samples_per_column=samples/width;
@@ -647,10 +697,10 @@ void audio_thumbnailer::init(int _channels,int _bits,int _samples) {
 	scale=1.0/offset;
 }
 int audio_thumbnailer::process_frame(uint8_t *_data,int samples_in_frame){
-	//begin by processing remaining samplesthread 4
-    //samples per column could be larger than a frame! (probably is)
+	//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 bytes=(bits>>3);
 	int stride=channels*bytes;
 	int in_sample=0;
 	while (in_sample<samples_in_frame&&column<width) {
-- 
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