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• Setting ffmpeg properly in ubuntu 16.04

  22 August 2017, by pro neon

  I am following this website for ffmpeg tutorial: http://dranger.com
  I tried to compile the programs after setting up ffmpeg in ubuntu by looking on some online videos but none of them worked. Some times GCC gives me undefined reference error and sometimes header not found error. I looked on some of the answers on SO that said that we need to do some change in the code as the new api is not backwards compatible but still GCC gives me undefined reference error.
  Here is the code that I am trying to compile:

      // tutorial01.c
  
  // Code based on a tutorial by Martin Bohme (boehme@inb.uni-luebeckREMOVETHIS.de)
  
  // Tested on Gentoo, CVS version 5/01/07 compiled with GCC 4.1.1
  
  // With updates from https://github.com/chelyaev/ffmpeg-tutorial
  
  // Updates tested on:
  
  // LAVC 54.59.100, LAVF 54.29.104, LSWS 2.1.101 
  
  // on GCC 4.7.2 in Debian February 2015
  
  
  
  // A small sample program that shows how to use libavformat and libavcodec to
  
  // read video from a file.
  
  //
  
  // Use
  
  //
  
  // gcc -o tutorial01 tutorial01.c -lavformat -lavcodec -lswscale -lz
  
  //
  
  // to build (assuming libavformat and libavcodec are correctly installed
  
  // your system).
  
  //
  
  // Run using
  
  //
  
  // tutorial01 myvideofile.mpg
  
  //
  
  // to write the first five frames from "myvideofile.mpg" to disk in PPM
  
  // format.
  
  
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libswscale></libswscale>swscale.h>
  
  
  
  #include 
  
  
  
  // compatibility with newer API
  
  #if LIBAVCODEC_VERSION_INT &lt; AV_VERSION_INT(55,28,1)
  
  #define av_frame_alloc avcodec_alloc_frame
  
  #define av_frame_free avcodec_free_frame
  
  #endif
  
  
  
  void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame) {
  
    FILE *pFile;
  
    char szFilename[32];
  
    int  y;
  
  
  
    // Open file
  
    sprintf(szFilename, "frame%d.ppm", iFrame);
  
    pFile=fopen(szFilename, "wb");
  
    if(pFile==NULL)
  
      return;
  
  
  
    // Write header
  
    fprintf(pFile, "P6\n%d %d\n255\n", width, height);
  
  
  
    // Write pixel data
  
    for(y=0; ydata[0]+y*pFrame->linesize[0], 1, width*3, pFile);
  
  
  
    // Close file
  
    fclose(pFile);
  
  }
  
  
  
  int main(int argc, char *argv[]) {
  
    // Initalizing these to NULL prevents segfaults!
  
    AVFormatContext   *pFormatCtx = NULL;
  
    int               i, videoStream;
  
    AVCodecContext    *pCodecCtxOrig = NULL;
  
    AVCodecContext    *pCodecCtx = NULL;
  
    AVCodec           *pCodec = NULL;
  
    AVFrame           *pFrame = NULL;
  
    AVFrame           *pFrameRGB = NULL;
  
    AVPacket          packet;
  
    int               frameFinished;
  
    int               numBytes;
  
    uint8_t           *buffer = NULL;
  
    struct SwsContext *sws_ctx = NULL;
  
  
  
    if(argc &lt; 2) {
  
      printf("Please provide a movie file\n");
  
      return -1;
  
    }
  
    // Register all formats and codecs
  
    av_register_all();
  
  
  
    // Open video file
  
    if(avformat_open_input(&amp;pFormatCtx, argv[1], NULL, NULL)!=0)
  
      return -1; // Couldn't open file
  
  
  
    // Retrieve stream information
  
    if(avformat_find_stream_info(pFormatCtx, NULL)&lt;0)
  
      return -1; // Couldn't find stream information
  
  
  
    // Dump information about file onto standard error
  
    av_dump_format(pFormatCtx, 0, argv[1], 0);
  
  
  
    // Find the first video stream
  
    videoStream=-1;
  
    for(i=0; inb_streams; i++)
  
      if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO) {
  
        videoStream=i;
  
        break;
  
      }
  
    if(videoStream==-1)
  
      return -1; // Didn't find a video stream
  
  
  
    // Get a pointer to the codec context for the video stream
  
    pCodecCtxOrig=pFormatCtx->streams[videoStream]->codec;
  
    // Find the decoder for the video stream
  
    pCodec=avcodec_find_decoder(pCodecCtxOrig->codec_id);
  
    if(pCodec==NULL) {
  
      fprintf(stderr, "Unsupported codec!\n");
  
      return -1; // Codec not found
  
    }
  
    // Copy context
  
    pCodecCtx = avcodec_alloc_context3(pCodec);
  
    if(avcodec_copy_context(pCodecCtx, pCodecCtxOrig) != 0) {
  
      fprintf(stderr, "Couldn't copy codec context");
  
      return -1; // Error copying codec context
  
    }
  
  
  
    // Open codec
  
    if(avcodec_open2(pCodecCtx, pCodec, NULL)&lt;0)
  
      return -1; // Could not open codec
  
  
  
    // Allocate video frame
  
    pFrame=av_frame_alloc();
  
  
  
    // Allocate an AVFrame structure
  
    pFrameRGB=av_frame_alloc();
  
    if(pFrameRGB==NULL)
  
      return -1;
  
  
  
    // Determine required buffer size and allocate buffer
  
    numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width,
  
                    pCodecCtx->height);
  
    buffer=(uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
  
  
  
    // Assign appropriate parts of buffer to image planes in pFrameRGB
  
    // Note that pFrameRGB is an AVFrame, but AVFrame is a superset
  
    // of AVPicture
  
    avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24,
  
           pCodecCtx->width, pCodecCtx->height);
  
  
  
    // initialize SWS context for software scaling
  
    sws_ctx = sws_getContext(pCodecCtx->width,
  
                 pCodecCtx->height,
  
                 pCodecCtx->pix_fmt,
  
                 pCodecCtx->width,
  
                 pCodecCtx->height,
  
                 PIX_FMT_RGB24,
  
                 SWS_BILINEAR,
  
                 NULL,
  
                 NULL,
  
                 NULL
  
                 );
  
  
  
    // Read frames and save first five frames to disk
  
    i=0;
  
    while(av_read_frame(pFormatCtx, &amp;packet)>=0) {
  
      // Is this a packet from the video stream?
  
      if(packet.stream_index==videoStream) {
  
        // Decode video frame
  
        avcodec_decode_video2(pCodecCtx, pFrame, &amp;frameFinished, &amp;packet);
  
  
  
        // Did we get a video frame?
  
        if(frameFinished) {
  
      // Convert the image from its native format to RGB
  
      sws_scale(sws_ctx, (uint8_t const * const *)pFrame->data,
  
            pFrame->linesize, 0, pCodecCtx->height,
  
            pFrameRGB->data, pFrameRGB->linesize);
  
  
  
      // Save the frame to disk
  
      if(++i&lt;=5)
  
        SaveFrame(pFrameRGB, pCodecCtx->width, pCodecCtx->height, 
  
              i);
  
        }
  
      }
  
  
  
      // Free the packet that was allocated by av_read_frame
  
      av_free_packet(&amp;packet);
  
    }
  
  
  
    // Free the RGB image
  
    av_free(buffer);
  
    av_frame_free(&amp;pFrameRGB);
  
  
  
    // Free the YUV frame
  
    av_frame_free(&amp;pFrame);
  
  
  
    // Close the codecs
  
    avcodec_close(pCodecCtx);
  
    avcodec_close(pCodecCtxOrig);
  
  
  
    // Close the video file
  
    avformat_close_input(&amp;pFormatCtx);
  
  
  
    return 0;
  
  }

  This is the command I use to compile:

  gcc -o tutorial01 tutorial01.c -lavformat -lavcodec -lswscale -lz

• Streaming RTP with ffmpeg and node.js to voip phone

  5 July 2023, by Nik Hendricks

  I am trying to implement SIP in node.js. Here is the library i am working on

  


  Upon receiving an invite request such as

  


  
Received INVITE
INVITE sip:201@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.39:5062;branch=z9hG4bK1534941205
From: "Nik" <sip:nik@192.168.1.2>;tag=564148403
To: <sip:201@192.168.1.2>
Call-ID: 2068254636@192.168.1.39
CSeq: 2 INVITE
Contact: <sip:nik@192.168.1.39:5062>
Authorization: Digest username="Nik", realm="NRegistrar", nonce="1234abcd", uri="sip:201@192.168.1.2:5060", response="7fba16dafe3d60c270b774bd5bba524c", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T42G 29.71.0.120
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 306

v=0
o=- 20083 20083 IN IP4 192.168.1.39
s=SDP data
c=IN IP4 192.168.1.39
t=0 0
m=audio 11782 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv



  


  I can then parse the SDP into an object like this

  


   
{
    "session":{
        "version":"0",
        "origin":"- 20084 20084 IN IP4 192.168.1.39",
        "sessionName":"SDP data"
    },
    "media":[
        {
            "media":"audio",
            "port":11784,
            "protocol":"RTP/AVP",
            "format":"0",
            "attributes":[
                "rtpmap:0 PCMU/8000",
                "rtpmap:8 PCMA/8000",
                "rtpmap:18 G729/8000",
                "fmtp:18 annexb=no",
                "rtpmap:9 G722/8000",
                "fmtp:101 0-15",
                "rtpmap:101 telephone-event/8000",
                "ptime:20",
                "sendrecv"
            ]
        }
    ]
}


  


  After sending the 100 and 180 responses with my library i attempt to start a RTP stream with ffmpeg

  


  var port = SDPParser.parse(res.message.body).media[0].port
var s = new STREAMER('output.wav', '192.168.1.39', port)


  


  with the following STREAMER class

  


  class Streamer{
    constructor(inputFilePath, rtpAddress, rtpPort){
        this.inputFilePath = 'output.wav';
        this.rtpAddress = rtpAddress;
        this.rtpPort = rtpPort;
    }

    start(){
        return new Promise((resolve) => {
            const ffmpegCommand = `ffmpeg -re -i ${this.inputFilePath} -ar 8000 -f mulaw -f rtp rtp://${this.rtpAddress}:${this.rtpPort}`;
            const ffmpegProcess = spawn(ffmpegCommand, { shell: true });
    
            ffmpegProcess.stdout.on('data', (data) => {
                data = data.toString()
                //replace all instances of 127.0.0.1 with our local ip address
                data = data.replace(new RegExp('127.0.0.1', 'g'), '192.168.1.3');

                resolve(data.toString())
            });
    
            ffmpegProcess.stderr.on('data', (data) => {
              // Handle stderr data if required
              console.log(data.toString())
            });
    
            ffmpegProcess.on('close', (code) => {
              // Handle process close event if required
              console.log('close')
              console.log(code.toString())
            });
    
            ffmpegProcess.on('error', (error) => {
              // Handle process error event if required
              console.log(error.toString())
            });
        })
    }
     
}


  


  the start() function resolves with the SDP that ffmpeg generates. I am starting to think that ffmpeg cant generate proper SDP for voip calls.

  


  so when i create 200 response with the following sdp

  


  v=0
o=- 0 0 IN IP4 192.168.1.3
s=Impact Moderato
c=IN IP4 192.168.1.39
t=0 0
a=tool:libavformat 58.29.100
m=audio 12123 RTP/AVP 97
b=AS:128
a=rtpmap:97 PCMU/8000/2


  


  the other line never picks up. from my understanding the first invite from the caller will provide SDP that will tell me where to send the RTP stream too and the correct codecs and everything. I know that currently, my wav file is PCMU and i can listen to it with ffplay and the provided sdp. what is required to make the other line pickup specifically a Yealink t42g

  


  my full attempt looks like this

  


  Client.on('INVITE', (res) => {
    console.log("Received INVITE")
    var d = Client.Dialog(res).then(dialog => {
        dialog.send(res.CreateResponse(100))
        dialog.send(res.CreateResponse(180))
        var port = SDPParser.parse(res.message.body).media[0].port

        var s = new STREAMER('output.wav', '192.168.1.39', port)
        s.start().then(sdp => {
            console.log(sdp.split('SDP:')[1])
            var ok = res.CreateResponse(200)
            ok.body = sdp.split('SDP:')[1]
            dialog.send(ok)
        })

        dialog.on('BYE', (res) => {
            console.log("BYE")
            dialog.send(res.CreateResponse(200))
            dialog.kill()
        })
    })
})


  


  I have provided a link to my library at the top of this message. My current problem is in the examples/Client folder.

  


  I'm not sure what could be going wrong here. Maybe i'm not using the right format or codec for the VOIP phone i dont see whats wrong with the SDP. especially if i can listen to SDP generated by ffmpeg if i stream RTP back to the same computer i use ffplay on. Any help is greatly appreciated.

  


  Update

  


  As i test i decided to send the caller back SDP that was generated by a Yealink phone like itself. but with some modifications

  


  v=0
o=- ${this.output_port} ${this.output_port} IN IP4 192.168.1.39
s=SDP data
c=IN IP4 192.168.1.39
t=0 0
m=audio ${this.output_port} RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:1
01 telephone-event/8000
a=ptime:20
a=sendrecv


  


  Finally, the phone that makes the call in the first place will fully answer but still no audio stream. I notice if I change the IP address or port to something wrong the other phone Will hear its own audio instead of just quiet. so this leads me to believe I am headed in the right direction. And maybe the problem lies in not sending the right audio format for what I'm describing.

  


  Additionaly, Whenever using ffmpeg to stream my audio with rtp I notice that it sees the file format as this pcm_alaw, 8000 Hz, mono, s16, 64 kb/s My new SDP describes using both ulaw and alaw but I'm not sure which it is saying it prefers

  


  v=0
o=- ${this.output_port} ${this.output_port} IN IP4 192.168.1.39
s=SDP data
c=IN IP4 192.168.1.39
t=0 0
m=audio ${this.output_port} RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:0
a=sendrecv


  


  I have been able to simply the SDP down to this. This will let the other phone actually pickup and not hear its own audio. it's just a completely dead air stream.

  


• H264 Encoding - Could not play video using VLC Player

  31 March 2016, by bot1131357

  I am have trouble encoding an H264 video correctly using FFmpeg libav. I could not play the encoded video in VLC media player, and although I could play the video on MPC-HC the time shows 00:00/00:00. Clearly I’m missing something.

  The Media info from MPC-HC shows this:

  General
  Format : AVC
  Format/Info : Advanced Video Codec
  File size : 110 KiB
  Duration : 2s 400ms
  Overall bit rate : 375 Kbps
  Writing library : x264 core 148 r2665 a01e339
  Encoding settings : cabac=0 / ref=3 / deblock=1:0:0 / analyse=0x1:0x111 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=7 / lookahead_threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=0 / weightp=0 / keyint=12 / keyint_min=1 / scenecut=40 / intra_refresh=0 / rc_lookahead=12 / rc=abr / mbtree=1 / bitrate=2000 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00

  Video
  Format : AVC
  Format/Info : Advanced Video Codec
  Format profile : Baseline@L2.1
  Format settings, CABAC : No
  Format settings, ReFrames : 3 frames
  Format settings, GOP : M=1, N=12
  Duration : 2s 400ms
  Bit rate : 2 000 Kbps
  Width : 320 pixels
  Height : 240 pixels
  Display aspect ratio : 4:3
  Frame rate mode : Variable
  Frame rate : 20.833 fps
  Color space : YUV
  Chroma subsampling : 4:2:0
  Bit depth : 8 bits
  Scan type : Progressive
  Bits/(Pixel*Frame) : 1.250
  Stream size : 586 KiB
  Writing library : x264 core 148 r2665 a01e339
  Encoding settings : cabac=0 / ref=3 / deblock=1:0:0 / analyse=0x1:0x111 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=7 / lookahead_threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=0 / weightp=0 / keyint=12 / keyint_min=1 / scenecut=40 / intra_refresh=0 / rc_lookahead=12 / rc=abr / mbtree=1 / bitrate=2000 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00

  I noticed something odd in the above info:
   The frame rate is 20.833 fps, instead of the specified 10 fps.
   Duration of 2s 400ms did not seem right either, since the video played for more than 4s.

  Also, (AVFrame* picture)->pict_type is always set to AV_PICTURE_TYPE_NONE. I don’t think this is normal.

  The library that I’m using is ffmpeg-20160219-git-98a0053-win32-dev. I would really really appreciate if you could help me out of this confusion.

  /*
  
   * Video encoding example
  
   */
  
  char filename[] = "test.mp4";
  
  int main(int argc, char** argv)
  
  {
  
      AVCodec *codec = NULL;
  
      AVCodecContext *codecCtx= NULL;
  
      AVFormatContext *pFormatCtx = NULL;
  
      AVStream * pVideoStream = NULL;
  
      AVFrame *picture = NULL;
  
  
  
      int i, x, y,            //
  
          ret,                // Return value
  
          got_packet_ptr;     // Data encoded into packet
  
  
  
      printf("Video encoding\n");
  
  
  
      // Register all formats and codecs
  
      av_register_all();
  
  
  
      // allocate context
  
      pFormatCtx = avformat_alloc_context();
  
      memcpy(pFormatCtx->filename,filename,
  
          min(strlen(filename), sizeof(pFormatCtx->filename)));
  
  
  
      // guess format
  
      pFormatCtx->oformat = av_guess_format("h264", NULL, NULL);
  
      if (NULL==pFormatCtx->oformat)
  
      {
  
          cerr << "Could not guess output format" << endl;
  
          return -1;
  
      }   
  
  
  
      // Find the codec.
  
      codec = avcodec_find_encoder(pFormatCtx->oformat->video_codec);
  
      if (codec == NULL) {
  
          fprintf(stderr, "Codec not found\n");
  
          return -1;
  
      }
  
  
  
      // Set context
  
      int framerate = 10;
  
      codecCtx = avcodec_alloc_context3(codec);
  
      avcodec_get_context_defaults3(codecCtx, codec); 
  
      codecCtx->pix_fmt = AV_PIX_FMT_YUV420P;
  
      codecCtx->profile = FF_PROFILE_H264_BASELINE; 
  
      // Resolution must be a multiple of two.
  
      codecCtx->width  = 320;
  
      codecCtx->height = 240;
  
  
  
      codecCtx->bit_rate = 2000000;
  
      codecCtx->time_base.den = framerate;
  
      codecCtx->time_base.num = 1;
  
      codecCtx->gop_size = 12; // emit one intra frame every twelve frames at most
  
  
  
      // Open the codec.  
  
      if (avcodec_open2(codecCtx, codec, NULL) < 0) 
  
      {
  
          printf("Cannot open video codec\n");
  
          return -1;
  
      }
  
  
  
      // Add stream to pFormatCtx
  
      pVideoStream = avformat_new_stream(pFormatCtx, codec);
  
      if (!pVideoStream) 
  
      {
  
          printf("Cannot add new video stream\n");
  
          return -1;
  
      }
  
      pVideoStream->codec = codecCtx;
  
      pVideoStream->time_base.den = framerate;
  
      pVideoStream->time_base.num = 1;
  
  
  
      if (avio_open2(&pFormatCtx->pb, filename, AVIO_FLAG_WRITE, NULL, NULL) < 0) 
  
      {
  
          printf("Cannot open file\n");
  
          return -1;
  
      }
  
  
  
      // Write file header.
  
      avformat_write_header(pFormatCtx, NULL);
  
  
  
      // Create frame
  
      picture= av_frame_alloc();
  
      picture->format = codecCtx->pix_fmt;
  
      picture->width  = codecCtx->width;
  
      picture->height = codecCtx->height;
  
  
  
      int bufferImgSize = av_image_get_buffer_size(codecCtx->pix_fmt, codecCtx->width,
  
                      codecCtx->height,1);    
  
      av_image_alloc(picture->data, picture->linesize, codecCtx->width, codecCtx->height,                 codecCtx->pix_fmt, 32);
  
  
  
      AVPacket avpkt;
  
  
  
      /* encode 1 second of video */
  
      for(i=0;i<50;i++) 
  
      {
  
          /* prepare a dummy image */
  
          /* Y */
  
          for(y=0;yheight;y++)
  
          {
  
              for(x=0;xwidth;x++) 
  
              {
  
                  picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
  
              }
  
          }
  
          /* Cb and Cr */
  
          for(y=0;yheight/2;y++) 
  
          {
  
              for(x=0;xwidth/2;x++) 
  
              {
  
                  picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
  
                  picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
  
              }
  
          }
  
  
  
          // Get timestamp
  
          picture->pts = (float) i * (1000.0/(float)(codecCtx->time_base.den)) * 90; 
  
  
  
          // Encode frame to packet
  
          av_init_packet(&avpkt);
  
          got_packet_ptr = 0;
  
          int error = avcodec_encode_video2(codecCtx, &avpkt, picture, &got_packet_ptr);
  
          if (!error && got_packet_ptr > 0) 
  
          {
  
              // Write packet with frame.
  
              ret = (av_interleaved_write_frame(pFormatCtx, &avpkt) == 0);        
  
          }   
  
          av_packet_unref(&avpkt); 
  
      }
  
  
  
      // Flush remaining encoded data
  
      while(1)
  
      {
  
          av_init_packet(&avpkt);
  
          got_packet_ptr = 0;
  
          // Encode frame to packet.
  
          int error = avcodec_encode_video2(codecCtx, &avpkt, NULL, &got_packet_ptr);
  
          if (!error && got_packet_ptr > 0) 
  
          {
  
              // Write packet with frame.
  
              ret = (av_interleaved_write_frame(pFormatCtx, &avpkt) == 0);        
  
          } 
  
          else 
  
          {
  
              break;
  
          }
  
          av_packet_unref(&avpkt); 
  
      }
  
      av_write_trailer(pFormatCtx);
  
  
  
      av_packet_unref(&avpkt);
  
      av_frame_free(&picture);
  
  
  
      avcodec_close(codecCtx);
  
      av_free(codecCtx);
  
  
  
      cin.get();
  
  }