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• ffmpeg concat doesn't keep video speed/framerate ?

  28 août 2016, par Nick

  I have a bunch of small webm clips in directory vidchunks. These clips were generated using javascript MediaRecorder API.

  MediaRecorder code :

  var mRecorder;
  
  var chunks = [];
  
  navigator.mediaDevices.getUserMedia({
  
    audio: false,
  
    video: {
  
    width: 1280,
  
    height: 760,
  
    mozMediaSource: "screen",
  
    mediaSource: "screen"
  
  }
  
  }).then(function(stream) {
  
    mRecorder = new MediaRecorder(stream,{mimeType:"video/webm"});
  
    mRecorder.ondataavailable = function(event) {
  
      var blob = event.data;
  
      chunks.push(blob);
  
      var vidchunk = new Blob(chunks);
  
      // save to directory vid chunk.
  
    };
  
    mRecorder.start(5000);
  
  }).catch(function(error) {
  
    console.log(error.message);
  
  });

  Then, once I have a bunch of these 5 second clips, I merge them :

  ffmpeg -safe 0 \
  
  -f concat -i <(for f in `ls vidchunks/* | sort -V`; do echo file $f; done) \
  
  -c:v copy -r 30 -vsync drop test.webm

  When I play back these chunks individually, they play correctly. All 5 seconds of video are rendered smoothly.

  However, once I merge, some 5 second chunks are condensed to like 1 second. So the total video, which should be 50 seconds for 10 chunks, turns out to be like 38 seconds because three of the chunks got condensed to 1 second.

  Any ideas on how to fix this ?

  EDIT : I’ve tried :

  • without "-r 30 -vsync drop"
  • without "-c:v copy"

  No changes...

  EDIT2 (some ffprobe outputs) :

  ffprobe 1472343170-1.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343170-1.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  ffprobe 1472343245-16.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343245-16.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  ffprobe 1472343350-37.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343350-37.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  I also noticed that the merged video plays normal when the mouse is moving around on the screen, but when the mouse is not moving, it moves 5x faster.

  I know this because I have a timer on the screen that I’m recording, and I can see that the timer speeds up significantly when mouse isn’t moving but when mouse is moving, the timer moves at normal speed.

  EDIT3 :
  The DTS/PTS looks like resets after each file is merged :

  DTS -9223363446915256, next:6866667 st:0 invalid dropping
  
  PTS -9223363446915256, next:6866667 invalid dropping st:0
  
  DTS -9223363446915222, next:6900000 st:0 invalid dropping
  
  PTS -9223363446915222, next:6900000 invalid dropping st:0
  
  DTS -9223363446915188, next:6933333 st:0 invalid dropping
  
  PTS -9223363446915188, next:6933333 invalid dropping st:0
  
  [concat @ 0x1cde3e0] DTS -9223363446920184 < 5006 out of order
  
  DTS -9223363446920184, next:5033333 st:0 invalid dropping
  
  PTS -9223363446920184, next:5033333 invalid dropping st:0
  
  DTS -9223363446920056, next:5066667 st:0 invalid dropping
  
  PTS -9223363446920056, next:5066667 invalid dropping st:0
  
  DTS -9223363446919956, next:5100000 st:0 invalid dropping
  
  PTS -9223363446919956, next:5100000 invalid dropping st:0

  EDIT4 : Tried ffmpeg -i "$f" -y -c copy -fflags +genpts "$f" and then merge again.

  Works a lot better, but now other files are getting skipped.

  Here’s ffprobe from a file that’s skipped :

  Metadata:
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:01.17, start: 0.000000, bitrate: 428 kb/s
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  [webm @ 0x1c01940] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, webm, to '1472343225-12.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760 [SAR 1:1 DAR 32:19], q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)

  Here’s ffprobe from a file not skipped :

  Input #0, matroska,webm, from '1472343215-10.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:05.03, start: 0.000000, bitrate: 80 kb/s
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  [webm @ 0x1349940] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, webm, to '1472343215-10.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760 [SAR 1:1 DAR 32:19], q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)

• Anomalie #3825 (Nouveau) : Manque le champ "langue_choisie" dans la base "spip_syndic_articles"

  31 août 2016, par Eric Camus

  Système Windows + IIS, PHP 5.4 MySQL 5.0, SPIP 3.1 et 3.0.
  Quand on met un site avec syndication alors il pleut des erreurs dans le "mysql.log" et "spip.log" :
  

  2016-08-31 07:13:25 195.83.13.139 (pid 20516) :Pub:ERREUR : Erreur 1054 de mysql : Unknown column ’langue_choisie’ in ’field list’
  UPDATE `xxxx`.spip_syndic_articles
  SET lang=’fr’,langue_choisie=’oui’
  WHERE id_syndic_article=6427
  

• FFmpeg RTP : Bad packed header lengths

  13 septembre 2016, par bot1131357

  I use the av_sdp_create() to generate my SDP file but I think something’s wrong. Here is the output :

  Output #0, rtp, to 'rtp://127.0.0.1:8554':
  
      Stream #0:0: Audio: vorbis (libvorbis), 44100 Hz, stereo, fltp, 64 kb/s
  
  v=0
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=No Name
  
  c=IN IP4 127.0.0.1
  
  t=0 0
  
  a=tool:libavformat 57.25.101
  
  m=audio 8554 RTP/AVP 96
  
  b=AS:64
  
  a=rtpmap:96 vorbis/44100/2
  
  a=fmtp:96 configuration=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

  If I try to use the SDP file anyway, I get the following message :

  >ffplay -i test2.sdp -protocol_whitelist file,udp,rtp
  
  ffplay version N-78598-g98a0053 Copyright (c) 2003-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 18.100 / 55. 18.100
  
    libavcodec     57. 24.103 / 57. 24.103
  
    libavformat    57. 25.101 / 57. 25.101
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 34.100 /  6. 34.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  [NULL @ 05aea880] Bad packed header lengths (30,0,616,3793) f=0/0
  
  [vorbis @ 05aea880] Extradata missing.
  
  [sdp @ 05ad7ec0] Failed to open codec in av_find_stream_info
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

  Am I overlooking something important ? Am I going about this the wrong way ? Any help would be much appreciated.

  Here is my source :

  #include "stdafx.h"
  
  #include 
  
  extern "C"
  
  {
  
  #include <libavutil></libavutil>opt.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavutil></libavutil>channel_layout.h>
  
  #include <libavutil></libavutil>common.h>
  
  #include <libavutil></libavutil>imgutils.h>
  
  #include <libavutil></libavutil>mathematics.h>
  
  #include <libavutil></libavutil>samplefmt.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  }
  
  
  
  /* check that a given sample format is supported by the encoder */
  
  static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
  
  {
  
      const enum AVSampleFormat *p = codec->sample_fmts;
  
  
  
      while (*p != AV_SAMPLE_FMT_NONE) {
  
          if (*p == sample_fmt)
  
              return 1;
  
          p++;
  
      }
  
      return 0;
  
  }
  
  
  
  
  
  /* just pick the highest supported samplerate */
  
  static int select_sample_rate(AVCodec *codec)
  
  {
  
      const int *p;
  
      int best_samplerate = 0;
  
  
  
      if (!codec->supported_samplerates)
  
          return 44100;
  
  
  
      p = codec->supported_samplerates;
  
      while (*p) {
  
          best_samplerate = FFMAX(*p, best_samplerate);
  
          p++;
  
      }
  
      return best_samplerate;
  
  }
  
  
  
  /* select layout with the highest channel count */
  
  static int select_channel_layout(AVCodec *codec)
  
  {
  
      const uint64_t *p;
  
      uint64_t best_ch_layout = 0;
  
      int best_nb_channels = 0;
  
  
  
      if (!codec->channel_layouts)
  
          return AV_CH_LAYOUT_STEREO;
  
  
  
      p = codec->channel_layouts;
  
      while (*p) {
  
          int nb_channels = av_get_channel_layout_nb_channels(*p);
  
  
  
          if (nb_channels > best_nb_channels) {
  
              best_ch_layout = *p;
  
              best_nb_channels = nb_channels;
  
          }
  
          p++;
  
      }
  
      return best_ch_layout;
  
  }
  
  
  
  
  
  static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
  
  {
  
      /* rescale output packet timestamp values from codec to stream timebase */
  
      av_packet_rescale_ts(pkt, *time_base, st->time_base);
  
  
  
      /* Write the compressed frame to the media file. */
  
      return av_interleaved_write_frame(fmt_ctx, pkt);
  
  }
  
  
  
  /*
  
  * Audio encoding example
  
  */
  
  static void audio_encode_example(const char *filename)
  
  {
  
      AVPacket pkt;
  
      int i, j, k, ret, got_output;
  
      int buffer_size;
  
  
  
      uint16_t *samples;
  
      float t, tincr;
  
  
  
  
  
      AVCodec *outCodec = NULL;
  
      AVCodecContext *outCodecCtx = NULL;
  
      AVFormatContext *outFormatCtx = NULL;
  
      AVStream * outAudioStream = NULL;
  
      AVFrame *outFrame = NULL;
  
  
  
      ret = avformat_alloc_output_context2(&amp;outFormatCtx, NULL, "rtp", filename);
  
      if (!outFormatCtx || ret &lt; 0)
  
      {
  
          fprintf(stderr, "Could not allocate output context");
  
      }
  
  
  
      outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER | AVFMT_FLAG_FLUSH_PACKETS;
  
      outFormatCtx->max_interleave_delta = 1;
  
      outFormatCtx->oformat->audio_codec = AV_CODEC_ID_VORBIS;
  
  
  
      /* find the encoder */
  
      outCodec = avcodec_find_encoder(outFormatCtx->oformat->audio_codec);
  
      if (!outCodec) {
  
          fprintf(stderr, "Codec not found\n");
  
          exit(1);
  
      }
  
  
  
      outAudioStream = avformat_new_stream(outFormatCtx, outCodec);
  
      if (!outAudioStream)
  
      {
  
          fprintf(stderr, "Cannot add new audio stream\n");
  
          exit(1);
  
      }
  
  
  
      outAudioStream->id = outFormatCtx->nb_streams - 1;
  
      outAudioStream->time_base.den = 44100; // 44.100 kHz
  
      outAudioStream->time_base.num = 1;
  
  
  
      outCodecCtx = outAudioStream->codec;
  
      outCodecCtx->bit_rate = 64000;
  
      outCodecCtx->time_base.den = outAudioStream->time_base.den;
  
      outCodecCtx->time_base.num = outAudioStream->time_base.num;
  
  
  
  
  
      /* check that the encoder supports input */
  
      outCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
  
      if (!check_sample_fmt(outCodec, outCodecCtx->sample_fmt)) {
  
          fprintf(stderr, "Encoder does not support sample format %s",
  
              av_get_sample_fmt_name(outCodecCtx->sample_fmt));
  
          exit(1);
  
      }
  
  
  
      /* select other audio parameters supported by the encoder */
  
      outCodecCtx->sample_rate = select_sample_rate(outCodec);
  
      outCodecCtx->channel_layout = select_channel_layout(outCodec);
  
      outCodecCtx->channels = av_get_channel_layout_nb_channels(outCodecCtx->channel_layout);
  
  
  
      /* open it */
  
      if (avcodec_open2(outCodecCtx, outCodec, NULL) &lt; 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      av_dump_format(outFormatCtx, 0, filename, 1);
  
  
  
      char buff[2048] = { 0 };
  
      av_sdp_create(&amp;outFormatCtx, 1, buff, 1024);
  
      printf("%s", buff);
  
  
  
      (...SDP printed here...)
  
  
  
      ret = avio_open2(&amp;outFormatCtx->pb, filename, AVIO_FLAG_WRITE, NULL, NULL);
  
      ret = avformat_write_header(outFormatCtx, NULL);
  
      printf("ret = %d\n", ret);
  
      if (ret &lt;0)
  
          exit(1);
  
  
  
      /* frame containing input audio */
  
      outFrame = av_frame_alloc();
  
      if (!outFrame) {
  
          fprintf(stderr, "Could not allocate audio frame\n");
  
          exit(1);
  
      }
  
  
  
      outFrame->nb_samples = outCodecCtx->frame_size;
  
      outFrame->format = outCodecCtx->sample_fmt;
  
      outFrame->channel_layout = outCodecCtx->channel_layout;
  
  
  
      /* the codec gives us the frame size, in samples,
  
      * we calculate the size of the samples buffer in bytes */
  
      buffer_size = av_samples_get_buffer_size(NULL, outCodecCtx->channels, outCodecCtx->frame_size,
  
          outCodecCtx->sample_fmt, 0);
  
      if (buffer_size &lt; 0) {
  
          fprintf(stderr, "Could not get sample buffer size\n");
  
          exit(1);
  
      }
  
      samples = (uint16_t*)av_malloc(buffer_size);
  
      if (!samples) {
  
          fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
  
              buffer_size);
  
          exit(1);
  
      }
  
      /* setup the data pointers in the AVFrame */
  
      ret = avcodec_fill_audio_frame(outFrame, outCodecCtx->channels, outCodecCtx->sample_fmt,
  
          (const uint8_t*)samples, buffer_size, 0);
  
      if (ret &lt; 0) {
  
          fprintf(stderr, "Could not setup audio frame\n");
  
          exit(1);
  
      }
  
  
  
      /* encode a single tone sound */
  
      t = 0;
  
      int next_pts = 0;
  
      tincr = 2 * M_PI * 440.0 / outCodecCtx->sample_rate;
  
      for (i = 0; i &lt; 44000; i++) {
  
          av_init_packet(&amp;pkt);
  
          pkt.data = NULL; // packet data will be allocated by the encoder
  
          pkt.size = 0;
  
  
  
          for (j = 0; j &lt; outCodecCtx->frame_size; j++) {
  
              samples[2 * j] = (int)(sin(t) * 10000);
  
  
  
              for (k = 1; k &lt; outCodecCtx->channels; k++)
  
                  samples[2 * j + k] = samples[2 * j];
  
              t += tincr;
  
          }
  
  
  
          // Sets time stamp
  
          next_pts += outFrame->nb_samples;
  
          outFrame->pts = next_pts;
  
  
  
          /* encode the samples */
  
          ret = avcodec_encode_audio2(outCodecCtx, &amp;pkt, outFrame, &amp;got_output);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error encoding audio frame\n");
  
              exit(1);
  
          }
  
          if (got_output) {
  
              //fwrite(pkt.data, 1, pkt.size, f);
  
              //pkt.stream_index = pktidx++;
  
              write_frame(outFormatCtx, &amp;outCodecCtx->time_base, outAudioStream, &amp;pkt);
  
              av_packet_unref(&amp;pkt);
  
          }
  
  
  
      }
  
  
  
      /* get the delayed frames */
  
      for (got_output = 1; got_output; i++) {
  
          ret = avcodec_encode_audio2(outCodecCtx, &amp;pkt, NULL, &amp;got_output);
  
          if (ret &lt; 0) {
  
              fprintf(stderr, "Error encoding frame\n");
  
              exit(1);
  
          }
  
  
  
          if (got_output) {
  
              //fwrite(pkt.data, 1, pkt.size, f);
  
              pkt.pts = AV_NOPTS_VALUE;
  
              write_frame(outFormatCtx, &amp;outCodecCtx->time_base, outAudioStream, &amp;pkt);
  
              av_packet_unref(&amp;pkt);
  
          }
  
      }
  
  
  
      av_freep(&amp;samples);
  
      av_frame_free(&amp;outFrame);
  
      avcodec_close(outCodecCtx);
  
      av_free(outCodecCtx);
  
  }
  
  
  
  
  
  int main(int argc, char **argv)
  
  {
  
      const char *output;
  
  
  
      /* register all the codecs */
  
      //avcodec_register_all();
  
      av_register_all();
  
      avformat_network_init(); // for network streaming
  
  
  
      audio_encode_example("rtp://127.0.0.1:8554");
  
  
  
      return 0;
  
  }