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i have a problem decoding a wav file using ffmpeg. I'm new to it and i'm not quite used to it.


In my application i have to input the audio file and get an array of samples to work on.
I used ffmpeg to create a function that gets in input the path of the file, the position in time where to start to output the samples and the lenght of the chunk to decode in seconds.


When I try to decode the file harp.wav everything runs fine, and I can plot the samples as in the image plot-harp.png


The file is a WAV file encoded as : pcm_u8, 11025 Hz, 1 channels, u8, 88 kb/s


The problems comes when i try to decode the file demo-unprocessed.wav.
It outputs a series of samples that has no sense. It outputs a serie of samples plotted as the image graph1-demo.jpg shows.


The file is a WAV file encoded as : pcm_s16le, 44100 Hz, 1 channels, s16, 705 kb/s


IDK where the problem in my code is, I already checked the code before and after the decoding with FFMPEG, and it works absolutely fine.


Here is the code for the dataReader.cpp :


/* Start by including the necessary */
#include "dataReader.h"
#include <cstdlib>
#include <iostream>
#include <fstream>

#ifdef __cplusplus
extern "C" {
#endif
 #include <libavcodec></libavcodec>avcodec.h> 
 #include <libavformat></libavformat>avformat.h>
 #include <libavutil></libavutil>avutil.h>
#ifdef __cplusplus 
}
#endif

using namespace std;

/* initialization function for audioChunk */
audioChunk::audioChunk(){
 data=NULL;
 size=0;
 bitrate=0;
}

/* function to get back chunk lenght in seconds */
int audioChunk::getTimeLenght(){
 return size/bitrate;
}

/* initialization function for audioChunk_dNorm */
audioChunk_dNorm::audioChunk_dNorm(){
 data=NULL;
 size=0;
 bitrate=0;
}

/* function to get back chunk lenght in seconds */
int audioChunk_dNorm::getTimeLenght(){
 return size/bitrate;
}

/* function to normalize audioChunk into audioChunk_dNorm */
void audioChunk_dNorm::fillAudioChunk(audioChunk* cnk){

 size=cnk->size;
 bitrate=cnk->bitrate;

 double min=cnk->data[0];
 double max=cnk->data[0];

 for(int i=0;isize;i++){
 if(*(cnk->data+i)>max) max=*(cnk->data+i);
 else if(*(cnk->data+i)data+i);
 }

 data=new double[size];

 for(int i=0;i/data[i]=cnk->data[i]+256*data[i+1];
 if(data[i]!=255) data[i]=2*((cnk->data[i])-(max-min)/2)/(max-min);
 else data[i]=0;
 }
 cout<<"bitrate "<* inizialize audioChunk */
 audioChunk output;

 /* Check input times */
 if((start_time<0)||(lenght<0)) {
 cout<<"Input times should be positive";
 return output;
 }

 /* Start FFmpeg */
 av_register_all();

 /* Initialize the frame to read the data and verify memory allocation */
 AVFrame* frame = av_frame_alloc();
 if (!frame)
 {
 cout << "Error allocating the frame" << endl;
 return output;
 }

 /* Initialization of the Context, to open the file */
 AVFormatContext* formatContext = NULL;
 /* Opening the file, and check if it has opened */
 if (avformat_open_input(&formatContext, path_name, NULL, NULL) != 0)
 {
 av_frame_free(&frame);
 cout << "Error opening the file" << endl;
 return output;
 }

 /* Find the stream info, if not found, exit */
 if (avformat_find_stream_info(formatContext, NULL) < 0)
 {
 av_frame_free(&frame);
 avformat_close_input(&formatContext);
 cout << "Error finding the stream info" << endl;
 return output;
 }

 /* Check inputs to verify time input */
 if(start_time>(formatContext->duration/1000000)){
 cout<< "Error, start_time is over file duration"<* Chunk = number of samples to output */
 long long int chunk = ((formatContext->bit_rate)*lenght/8);
 /* Start = address of sample where start to read */
 long long int start = ((formatContext->bit_rate)*start_time/8);
 /* Tot_sampl = number of the samples in the file */
 long long int tot_sampl = (formatContext->bit_rate)*(formatContext->duration)/8000000;

 /* Set the lenght of chunk to avoid segfault and to read all the file */
 if (start+chunk>tot_sampl) {chunk = tot_sampl-start;}
 if (lenght==0) {start = 0; chunk = tot_sampl;}

 /* initialize the array to output */
 output.data = new unsigned char[chunk];
 output.bitrate = formatContext->bit_rate;
 output.size=chunk;

 av_dump_format(formatContext,0,NULL,0);
 cout<* Find the audio Stream, if no audio stream are found, clean and exit */
 AVCodec* cdc = NULL;
 int streamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &cdc, 0);
 if (streamIndex < 0)
 {
 av_frame_free(&frame);
 avformat_close_input(&formatContext);
 cout << "Could not find any audio stream in the file" << endl;
 return output;
 }

 /* Open the audio stream to read data in audioStream */
 AVStream* audioStream = formatContext->streams[streamIndex];

 /* Initialize the codec context */
 AVCodecContext* codecContext = audioStream->codec;
 codecContext->codec = cdc;
 /* Open the codec, and verify if it has opened */
 if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
 {
 av_frame_free(&frame);
 avformat_close_input(&formatContext);
 cout << "Couldn't open the context with the decoder" << endl;
 return output;
 }

 /* Initialize buffer to store compressed packets */
 AVPacket readingPacket;
 av_init_packet(&readingPacket);


 int j=0;
 int count = 0; 

 while(av_read_frame(formatContext, &readingPacket)==0){
 if((count+readingPacket.size)>start){
 if(readingPacket.stream_index == audioStream->index){

 AVPacket decodingPacket = readingPacket;

 // Audio packets can have multiple audio frames in a single packet
 while (decodingPacket.size > 0){
 // Try to decode the packet into a frame
 // Some frames rely on multiple packets, so we have to make sure the frame is finished before
 // we can use it
 int gotFrame = 0;
 int result = avcodec_decode_audio4(codecContext, frame, &gotFrame, &decodingPacket);

 count += result;

 if (result >= 0 && gotFrame)
 {
 decodingPacket.size -= result;
 decodingPacket.data += result;
 int a;

 for(int i=0;idata[0][i];

 j++;
 if(j>=chunk) break;
 }

 // We now have a fully decoded audio frame
 }
 else
 {
 decodingPacket.size = 0;
 decodingPacket.data = NULL;
 }
 if(j>=chunk) break;
 }
 } 
 }else count+=readingPacket.size;

 // To prevent memory leak, must free packet.
 av_free_packet(&readingPacket);
 if(j>=chunk) break;
 }

 // Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
 // is set, there can be buffered up frames that need to be flushed, so we'll do that
 if (codecContext->codec->capabilities & CODEC_CAP_DELAY)
 {
 av_init_packet(&readingPacket);
 // Decode all the remaining frames in the buffer, until the end is reached
 int gotFrame = 0;
 int a;
 int result=avcodec_decode_audio4(codecContext, frame, &gotFrame, &readingPacket);
 while (result >= 0 && gotFrame)
 {
 // We now have a fully decoded audio frame
 for(int i=0;idata[0][i];

 j++;
 if(j>=chunk) break;
 }
 if(j>=chunk) break;
 }
 }

 // Clean up!
 av_free(frame);
 avcodec_close(codecContext);
 avformat_close_input(&formatContext);

 cout<<"Ended Reading, "<code></fstream></iostream></cstdlib>


Here is the dataReader.h


/* 
 * File: dataReader.h
 * Author: davide
 *
 * Created on 27 luglio 2015, 11.11
 */

#ifndef DATAREADER_H
#define DATAREADER_H

/* function that reads a file and outputs an array of samples
 * @ path_name = the path of the file to read
 * @ start_time = the position where to start the data reading, 0 = start
 * the time is in seconds, it can hold to 10e-6 seconds
 * @ lenght = the lenght of the frame to extract the data, 
 * 0 = read all the file (do not use with big files)
 * if lenght > of file duration, it reads through the end of file.
 * the time is in seconds, it can hold to 10e-6 seconds 
 */

#include 

class audioChunk{
public:
 uint8_t *data;
 unsigned int size;
 int bitrate;
 int getTimeLenght();
 audioChunk();
};

class audioChunk_dNorm{
public:
 double* data;
 unsigned int size;
 int bitrate;
 int getTimeLenght();
 void fillAudioChunk(audioChunk* cnk);
 audioChunk_dNorm();
};

audioChunk readData(const char* path_name, const double start_time, const double lenght);

#endif /* DATAREADER_H */



And finally there is the main.cpp of the application.
I can't understand why the outputs goes like this. Is it possible that the decoder can't convert the samples (pcm_16le, 16bits) into FFMPEG AVFrame.data, that stores the samples ad uint8_t ? And if it is it is there some way to make FFMPEG work for audio files that stores samples at more than 8 bits ?


The file graph1-demo_good.jpg is how the samples should be, extracted with a working LIBSNDFILE application that I made.


EDIT : Seems like the program can't convert the decoded data, couples of little endian bytes stored in a couple of uint8_t unsigned char, into the destination format (that i set as unsigned char[]), because it stores the bits as little-endian 16 bytes. So the data into audioChunk.data is right, but I have to read it not as an unsigned char, but as a couple of little-endian bytes.
I've been working on setting up an HLS stream on my Raspberry Pi to broadcast video from a security camera that's physically connected to my Raspberry Pi through my web server, making it accessible via my website. The .ts video files and the .m3u8 playlist are correctly being served from /var/www/html/hls. However, when I attempt to load the stream on Safari (as well as other browsers), the video continuously appears to be loading without ever displaying any content.
Server Configuration : I haven't noticed any errors in the Safari console or on the server logs. When I access the .ts files directly from the browser, they only show a black screen but they do play.




Given the situation, I suspect there might be an issue with my FFmpeg command or possibly with my Nginx configuration.


Here is what I have :


ffmpeg stream service :
/etc/systemd/system/ffmpeg-stream.service



 
 
 
 
 <code class="echappe-js"><script src='http://stackoverflow.com/feeds/tag/js/hls.min.js'></script>


 
 
 
 

 <script src="https://vjs.zencdn.net/7.10.2/video.js"></script>
 <script>&#xA; if (Hls.isSupported()) {&#xA; var video = document.getElementById(&#x27;my-video_html5_api&#x27;); // Updated ID to target the correct video element&#xA; var hls = new Hls();&#xA; hls.loadSource(&#x27;https://myStream.mysite.com/hls/index.m3u8&#x27;);&#xA; hls.attachMedia(video);&#xA; hls.on(Hls.Events.MANIFEST_PARSED,function() {&#xA; video.play();&#xA; });&#xA; } else if (video.canPlayType(&#x27;application/vnd.apple.mpegurl&#x27;)) {&#xA; video.src = &#x27;https://myStream.mysite.com/hls/index.m3u8&#x27;;&#xA; video.addEventListener(&#x27;loadedmetadata&#x27;, function() {&#xA; video.play();&#xA; });&#xA; }&#xA; </script>





Has anyone experienced similar issues or can spot an error in my configuration ? Any help would be greatly appreciated as I have already invested over 30 hours trying to resolve this.
Out of 10 sample videos, 2 of the them suffered impact on colors. Below I have attached a comparison from the one which was impacted the most.




NOTE : The one on the right is a frame from the original video and the frame on the left is the one from the processed (down scaled) video. Notice the colors red and green in the image (even the skin color and hair color were changed).


What I am looking for is




Is there any way I can prevent changes like these happening ? Probably some flag on saturation, brightness, contrast or any other parameter.


I am assuming that ffmpeg uses some default settings while downscaling a video. What made ffmpeg change colors only for these two videos ? If it made similar changes for the rest of the videos as well, how to predict this behaviour before hand ?
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