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• Encoding using ffmpeg library fails

  29 septembre 2012, par Erik Swansson

  I've spent some time looking at the ffmpeg library and setting things up, I'm opening a .flv file. Reading and decoding and the frames, now I'm trying to encode it to MP4 but my packets end up empty.

  My code as follows

  int main (){
  
  
  
      avformat_open_input(&pFC, "c://wav//test2.flv", NULL, NULL);
  
  
  
      po = av_find_stream_info(pFC);
  
  
  
      //ADD LOGIC TO FIND VIDEO STREAM
  
      pCodecC = pFC->streams[0]->codec;
  
  
  
      decoder = avcodec_find_decoder(pCodecC->codec_id);
  
      encoder = avcodec_find_encoder(pCodecC->codec_id);
  
      po = avcodec_open(pCodecC, decoder);
  
  
  
      pCodecE =  avcodec_alloc_context3(encoder);
  
      /* put sample parameters */
  
      pCodecE->bit_rate = 400000;
  
      /* resolution must be a multiple of two */
  
      pCodecE->width = 352;
  
      pCodecE->height = 288;
  
      /* frames per second */
  
      pCodecE->time_base.den = 25;
  
      pCodecE->time_base.num = 1;
  
      pCodecE->gop_size = 10; /* emit one intra frame every ten frames */
  
      pCodecE->max_b_frames=1;
  
      pCodecE->pix_fmt = PIX_FMT_YUV420P;
  
  
  
      if(pCodecC->codec_id == CODEC_ID_H264)
  
          av_opt_set(pCodecE->priv_data, "preset", "slow", 0);
  
  
  
      po =  avcodec_open2(pCodecE, encoder, NULL);
  
  
  
      AVFrame *pFrame;
  
      // Allocate an AVFrame structure
  
  
  
  
  
  
  
      // Allocate video frame
  
      pFrame=avcodec_alloc_frame();
  
      int frameFinished = 0;
  
      int frame = 0;
  
      int gotpacket = 0;
  
  
  
      while(av_read_frame(pFC, &packet) >= 0)
  
      {
  
          if(packet.stream_index==0) //the video stream is 0
  
          {
  
              int len = avcodec_decode_video2(pCodecC, pFrame, &frameFinished, &packet);
  
              if(frameFinished)
  
              {
  
                  printf("frame # %i", frame);
  
  
  
                  po =avcodec_encode_video2(pCodecE, &spacket, pFrame, &gotpacket);
  
                  if(gotpacket)
  
                  {
  
                      printf("packet recieved");
  
                  }
  
                  frame++;
  
              }
  
          }
  
          av_free_packet(&packet);
  
      }
  
  
  
  
  
      printf("encoding done");
  
  
  
      return 0;
  
  }

  Basically everything works up to

   po =avcodec_encode_video2(pCodecE, &spacket, pFrame, &gotpacket);

  Where &gotpacket returns 0, as in an empty frame.

  Not sure what I'm doing wrong.

• Sending raw h264 video and aac audio frames to an RTMP server using ffmpeg

  15 août 2016, par codeimpaler

  I am receiving raw h264 and aac audio frames from an even driven source. I am trying to send these frames to an rtmp server.
  I started working from the ffmpeg example muxing.c which successfully sends a custom stream to the rtmp server. I figure I just need to replace their frame data with my own.I found this suggestion online. I have tried How to pack raw h264 stream to flv container and send over rtmp using ffmpeg (not command)
  and
  How to publish selfmade stream with ffmpeg and c++ to rtmp server ?
  and a few other suggestions but none have worked for me.
  I have tried to directly memcpy my byte buffer but my code keeps failing
  at ret = avcodec_encode_video2(c, &pkt, frame, &got_packet).
  Specifically, I get an invalid access error.
  For a little more context, anytime I receive a frame (which is event driven), void RTMPWriter::WriteVideoFrame(...) is called. Assume the constructor has already been called before the first frame is received.
  I am not that familiar with ffmpeg and there could be several things wrong with the code. Any input will be really appreciated.

      #define STREAM_FRAME_RATE 25 /* 25 images/s */
  
      #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
  
      #define SCALE_FLAGS SWS_BICUBIC
  
      RTMPWriter::RTMPWriter()
  
        : seenKeyFrame(false),
  
          video_st({ 0 }), 
  
          audio_st({ 0 }),
  
          have_video(0), 
  
          have_audio(0)
  
      {
  
  
  
          const char *filename;
  
          AVCodec *audio_codec = NULL, *video_codec = NULL;
  
          int ret;
  
  
  
          int encode_video = 0, encode_audio = 0;
  
          AVDictionary *opt = NULL;
  
          int i;
  
  
  
          /* Initialize libavcodec, and register all codecs and formats. */
  
          av_register_all();
  
  
  
          avformat_network_init();
  
  
  
         String^ StreamURL = "StreamURL";
  
         String^ out_uri = safe_cast(ApplicationData::Current->LocalSettings->Values->Lookup(StreamURL));
  
         std::wstring out_uriW(out_uri->Begin());
  
         std::string out_uriA(out_uriW.begin(), out_uriW.end());
  
         filename = out_uriA.c_str();  
  
  
  
         /* allocate the output media context */
  
         avformat_alloc_output_context2(&oc, NULL, "flv", filename);
  
         if (!oc)
  
         {
  
             OutputDebugString(L"Could not deduce output format from file extension: using MPEG.\n");
  
             avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
  
         }
  
         if (!oc)
  
         {
  
             OutputDebugString(L"Could not allocate  using MPEG.\n");
  
         }
  
  
  
  
  
         fmt = oc->oformat;
  
  
  
         /* Add the audio and video streams using the default format codecs
  
         * and initialize the codecs. */
  
         if (fmt->video_codec != AV_CODEC_ID_NONE) {
  
             add_stream(&video_st, oc, &video_codec, fmt->video_codec);
  
             have_video = 1;
  
             encode_video = 1;
  
         }
  
         if (fmt->audio_codec != AV_CODEC_ID_NONE) {
  
             add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
  
             have_audio = 1;
  
             encode_audio = 1;
  
         }
  
  
  
         /* Now that all the parameters are set, we can open the audio and
  
          * video codecs and allocate the necessary encode buffers. */
  
         if (have_video)
  
         {
  
             open_video(oc, video_codec, &video_st, opt);
  
         }
  
  
  
         if (have_audio)
  
         {
  
             open_audio(oc, audio_codec, &audio_st, opt);
  
         }
  
  
  
         av_dump_format(oc, 0, filename, 1);
  
  
  
         /* open the output file, if needed */
  
         if (!(fmt->flags & AVFMT_NOFILE))
  
         {
  
             ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
  
             if (ret < 0)
  
             {
  
                 OutputDebugString(L"Could not open ");
  
                 OutputDebugString(out_uri->Data());
  
             }
  
         }
  
  
  
         /* Write the stream header, if any. */
  
         ret = avformat_write_header(oc, &opt);
  
         if (ret < 0)
  
         {
  
             OutputDebugString(L"Error occurred when writing stream header \n");
  
         }
  
  
  
      }
  
  
  
      void RTMPWriter::WriteVideoFrame(
  
          boolean isKeyFrame,
  
          boolean hasDiscontinuity,
  
          UINT64 frameId,
  
          UINT32 videoBufferLength,
  
          BYTE *videoBytes)
  
      {
  
  
  
          int ret;
  
          AVCodecContext *c;
  
          AVFrame* frame;
  
          int got_packet = 0;
  
          AVPacket pkt = { 0 };
  
  
  
          c = video_st.enc;
  
  
  
          frame = get_video_frame(videoBufferLength, videoBytes);
  
  
  
          /* encode the image */
  
          ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
  
          if (ret < 0) {
  
               OutputDebugString(L"Error encoding video frame: \n")
  
          }
  
  
  
          if (got_packet) 
  
          {
  
              ret = write_frame(oc, &c->time_base, video_st.st, &pkt);
  
          }
  
          else {
  
              ret = 0;
  
          }
  
  
  
          if (ret < 0) {
  
               OutputDebugString(L"Error while writing video frame: %s\n");
  
          }
  
      }
  
  
  
      AVFrame * RTMPWriter::get_video_frame(
  
         UINT32 videoBufferLength,
  
         BYTE *videoBytes)
  
      {
  
          AVCodecContext *c = video_st.enc;
  
  
  
          if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
  
              /* as we only generate a YUV420P picture, we must convert it
  
              * to the codec pixel format if needed */
  
              if (!video_st.sws_ctx) {
  
                  video_st.sws_ctx = sws_getContext(c->width, c->height,
  
                      AV_PIX_FMT_YUV420P,
  
                      c->width, c->height,
  
                      c->pix_fmt,
  
                      SCALE_FLAGS, NULL, NULL, NULL);
  
                  if (!video_st.sws_ctx) {
  
                      fprintf(stderr,
  
                          "Could not initialize the conversion context\n");
  
                          exit(1);
  
                  }
  
              }
  
              fill_yuv_image(video_st.tmp_frame, video_st.next_pts, c->width, c->height, videoBufferLength, videoBytes);
  
              sws_scale(video_st.sws_ctx,
  
              (const uint8_t * const *)video_st.tmp_frame->data, video_st.tmp_frame->linesize,
  
              0, c->height, video_st.frame->data, video_st.frame->linesize);
  
          }
  
          else {
  
              fill_yuv_image(video_st.frame, video_st.next_pts, c->width, c->height, videoBufferLength, videoBytes);
  
          }
  
  
  
          video_st.frame->pts = video_st.next_pts++;
  
  
  
          return video_st.frame;
  
      }
  
  
  
      /* Prepare a dummy image. */
  
      void  RTMPWriter::fill_yuv_image(
  
           AVFrame *pict, 
  
           int frame_index,
  
           int width, 
  
           int height, 
  
           UINT32 videoBufferLength,
  
           BYTE *videoBytes)
  
      {
  
          //int x, y, i, ret;
  
  
  
          /* when we pass a frame to the encoder, it may keep a reference to it
  
          * internally;
  
          * make sure we do not overwrite it here
  
          */
  
          ret = av_frame_make_writable(pict);
  
          if (ret < 0) 
  
          {
  
               OutputDebugString(L"Unable to make piture writable");
  
          }
  
  
  
          memcpy(pict->data, videoBytes, videoBufferLength);
  
  
  
          //i = frame_index;
  
  
  
          ///* Y */
  
          //for (y = 0; y < height; y++)
  
          //  for (x = 0; x < width; x++)
  
          //      pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
  
  
  
          ///* Cb and Cr */
  
          //for (y = 0; y < height / 2; y++) {
  
          //  for (x = 0; x < width / 2; x++) {
  
          //      pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
  
          //      pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
  
          //  }
  
          //}
  
      }
  
  
  
      void RTMPWriter::WriteAudioFrame()
  
      {
  
  
  
      }
  
  
  
      /* Add an output stream. */
  
      void  RTMPWriter::add_stream(
  
          OutputStream *ost, 
  
          AVFormatContext *oc,
  
          AVCodec **codec,
  
          enum AVCodecID codec_id)
  
     {
  
      AVCodecContext *c;
  
      int i;
  
  
  
      /* find the encoder */
  
      *codec = avcodec_find_encoder(codec_id);
  
      if (!(*codec)) {
  
          OutputDebugString(L"Could not find encoder for '%s'\n");
  
          //avcodec_get_name(codec_id));
  
          exit(1);
  
      }
  
  
  
      ost->st = avformat_new_stream(oc, NULL);
  
      if (!ost->st) {
  
          OutputDebugString(L"Could not allocate stream\n");
  
          exit(1);
  
      }
  
      ost->st->id = oc->nb_streams - 1;
  
      c = avcodec_alloc_context3(*codec);
  
      if (!c) {
  
          OutputDebugString(L"Could not alloc an encoding context\n");
  
          exit(1);
  
      }
  
      ost->enc = c;
  
  
  
      switch ((*codec)->type) {
  
      case AVMEDIA_TYPE_AUDIO:
  
          c->sample_fmt = (*codec)->sample_fmts ?
  
              (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
  
          c->bit_rate = 64000;
  
          c->sample_rate = 44100;
  
          if ((*codec)->supported_samplerates) {
  
              c->sample_rate = (*codec)->supported_samplerates[0];
  
              for (i = 0; (*codec)->supported_samplerates[i]; i++) {
  
                  if ((*codec)->supported_samplerates[i] == 44100)
  
                      c->sample_rate = 44100;
  
              }
  
          }
  
          c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
  
          c->channel_layout = AV_CH_LAYOUT_STEREO;
  
          if ((*codec)->channel_layouts) {
  
              c->channel_layout = (*codec)->channel_layouts[0];
  
              for (i = 0; (*codec)->channel_layouts[i]; i++) {
  
                  if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
  
                      c->channel_layout = AV_CH_LAYOUT_STEREO;
  
              }
  
          }
  
          c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
  
          ost->st->time_base = /*(AVRational)*/{ 1, c->sample_rate };
  
          break;
  
  
  
      case AVMEDIA_TYPE_VIDEO:
  
          c->codec_id = codec_id;
  
  
  
          c->bit_rate = 400000;
  
          /* Resolution must be a multiple of two. */
  
          c->width = 352;
  
          c->height = 288;
  
          /* timebase: This is the fundamental unit of time (in seconds) in terms
  
          * of which frame timestamps are represented. For fixed-fps content,
  
          * timebase should be 1/framerate and timestamp increments should be
  
          * identical to 1. */
  
          ost->st->time_base = /*(AVRational)*/{ 1, STREAM_FRAME_RATE };
  
          c->time_base = ost->st->time_base;
  
  
  
          c->gop_size = 12; /* emit one intra frame every twelve frames at most */
  
          c->pix_fmt = STREAM_PIX_FMT;
  
              if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
  
                  /* just for testing, we also add B-frames */
  
                  c->max_b_frames = 2;
  
              }
  
              if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
  
                  /* Needed to avoid using macroblocks in which some coeffs overflow.
  
                  * This does not happen with normal video, it just happens here as
  
                  * the motion of the chroma plane does not match the luma plane. */
  
                  c->mb_decision = 2;
  
              }
  
              break;
  
  
  
          default:
  
              break;
  
          }
  
  
  
           /* Some formats want stream headers to be separate. */
  
          if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
              c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
  
      }
  
  
  
  AVFrame * RTMPWriter::alloc_audio_frame(
  
      enum AVSampleFormat sample_fmt,
  
      uint64_t channel_layout,
  
      int sample_rate, int nb_samples)
  
  {
  
      AVFrame *frame = av_frame_alloc();
  
      int ret;
  
  
  
      if (!frame) {
  
          OutputDebugString(L"Error allocating an audio frame\n");
  
          exit(1);
  
      }
  
  
  
      frame->format = sample_fmt;
  
      frame->channel_layout = channel_layout;
  
      frame->sample_rate = sample_rate;
  
      frame->nb_samples = nb_samples;
  
  
  
      if (nb_samples) {
  
          ret = av_frame_get_buffer(frame, 0);
  
          if (ret < 0) {
  
              OutputDebugString(L"Error allocating an audio buffer\n");
  
              exit(1);
  
          }
  
      }
  
  
  
          return frame;
  
      }
  
  
  
  
  
  
  
  
  
  void  RTMPWriter::open_audio(
  
      AVFormatContext *oc, 
  
      AVCodec *codec, 
  
      OutputStream *ost, 
  
      AVDictionary *opt_arg)
  
  {
  
      AVCodecContext *c;
  
      int nb_samples;
  
      int ret;
  
      AVDictionary *opt = NULL;
  
  
  
      c = ost->enc;
  
  
  
      /* open it */
  
      av_dict_copy(&opt, opt_arg, 0);
  
      ret = avcodec_open2(c, codec, &opt);
  
      av_dict_free(&opt);
  
      if (ret < 0) {
  
          OutputDebugString(L"Could not open audio codec: %s\n");// , av_err2str(ret));
  
          exit(1);
  
      }
  
  
  
      /* init signal generator */
  
      ost->t = 0;
  
      ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
  
      /* increment frequency by 110 Hz per second */
  
      ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
  
  
  
      if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
  
          nb_samples = 10000;
  
      else
  
          nb_samples = c->frame_size;
  
  
  
      ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
  
          c->sample_rate, nb_samples);
  
      ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
  
          c->sample_rate, nb_samples);
  
  
  
      /* copy the stream parameters to the muxer */
  
      ret = avcodec_parameters_from_context(ost->st->codecpar, c);
  
      if (ret < 0) {
  
          OutputDebugString(L"Could not copy the stream parameters\n");
  
          exit(1);
  
      }
  
  
  
      /* create resampler context */
  
      ost->swr_ctx = swr_alloc();
  
      if (!ost->swr_ctx) {
  
          OutputDebugString(L"Could not allocate resampler context\n");
  
          exit(1);
  
      }
  
  
  
      /* set options */
  
      av_opt_set_int(ost->swr_ctx, "in_channel_count", c->channels, 0);
  
      av_opt_set_int(ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
  
      av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
  
      av_opt_set_int(ost->swr_ctx, "out_channel_count", c->channels, 0);
  
      av_opt_set_int(ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
  
      av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
  
  
  
      /* initialize the resampling context */
  
      if ((ret = swr_init(ost->swr_ctx)) < 0) {
  
          OutputDebugString(L"Failed to initialize the resampling context\n");
  
          exit(1);
  
      }
  
  }
  
  
  
  int RTMPWriter::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);
  
      pkt->stream_index = st->index;
  
  
  
      /* Write the compressed frame to the media file. */
  
      //log_packet(fmt_ctx, pkt);
  
      OutputDebugString(L"Actually sending video frame: %s\n");
  
      return av_interleaved_write_frame(fmt_ctx, pkt);
  
  }
  
  
  
  
  
  AVFrame  *RTMPWriter::alloc_picture(
  
      enum AVPixelFormat pix_fmt, 
  
      int width, 
  
      int height)
  
  {
  
      AVFrame *picture;
  
      int ret;
  
  
  
      picture = av_frame_alloc();
  
      if (!picture)
  
          return NULL;
  
  
  
      picture->format = pix_fmt;
  
      picture->width = width;
  
      picture->height = height;
  
  
  
      /* allocate the buffers for the frame data */
  
      ret = av_frame_get_buffer(picture, 32);
  
      if (ret < 0) {
  
          fprintf(stderr, "Could not allocate frame data.\n");
  
          exit(1);
  
      }
  
  
  
      return picture;
  
  }
  
  
  
  void RTMPWriter::open_video(
  
      AVFormatContext *oc, 
  
      AVCodec *codec, 
  
      OutputStream *ost, 
  
      AVDictionary *opt_arg)
  
  {
  
      int ret;
  
      AVCodecContext *c = ost->enc;
  
      AVDictionary *opt = NULL;
  
  
  
      av_dict_copy(&opt, opt_arg, 0);
  
  
  
      /* open the codec */
  
      ret = avcodec_open2(c, codec, &opt);
  
      av_dict_free(&opt);
  
      if (ret < 0) {
  
          OutputDebugString(L"Could not open video codec: %s\n");// , av_err2str(ret));
  
          exit(1);
  
      }
  
  
  
      /* allocate and init a re-usable frame */
  
      ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
  
      if (!ost->frame) {
  
          OutputDebugString(L"Could not allocate video frame\n");
  
          exit(1);
  
      }
  
  
  
      /* If the output format is not YUV420P, then a temporary YUV420P
  
      * picture is needed too. It is then converted to the required
  
      * output format. */
  
      ost->tmp_frame = NULL;
  
      if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
  
          ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
  
          if (!ost->tmp_frame) {
  
              OutputDebugString(L"Could not allocate temporary picture\n");
  
              exit(1);
  
          }
  
      }
  
  
  
      /* copy the stream parameters to the muxer */
  
      ret = avcodec_parameters_from_context(ost->st->codecpar, c);
  
      if (ret < 0) {
  
          OutputDebugString(L"Could not copy the stream parameters\n");
  
          exit(1);
  
      }
  
  }
  
  
  
  void RTMPWriter::close_stream(AVFormatContext *oc, OutputStream *ost)
  
  {
  
      avcodec_free_context(&ost->enc);
  
      av_frame_free(&ost->frame);
  
      av_frame_free(&ost->tmp_frame);
  
      sws_freeContext(ost->sws_ctx);
  
      swr_free(&ost->swr_ctx);
  
  }
  
  
  
  RTMPWriter::~RTMPWriter()
  
  {
  
      av_write_trailer(oc);
  
      /* Close each codec. */
  
      if (have_video)
  
          close_stream(oc, &video_st);
  
      if (have_audio)
  
          close_stream(oc, &audio_st);
  
  
  
      if (!(fmt->flags & AVFMT_NOFILE))
  
          /* Close the output file. */
  
          avio_closep(&oc->pb);
  
  
  
      /* free the stream */
  
      avformat_free_context(oc);
  
  }

• FFmpeg RTP payload 96 instead of 97

  26 octobre 2016, par bot1131357

  I am trying to create an rtp audio stream with ffmpeg. The application output and SDP file configuration are as follows :

  Output #0, rtp, to 'rtp://127.0.0.1:8554':
  
      Stream #0:0: Audio: pcm_s16be, 8000 Hz, stereo, s16, 256 kb/s
  
  
  
  SDP:    
  
  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:256
  
  a=rtpmap:96 L16/8000/2

  However, when I try to read it with ffplay -i test.sdp -protocol_whitelist file,udp,rtp, it fails,shows the following :

  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
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0
  
      (...waits indefinitely.)

  The only way to make it work again is to modify the payload type in the SDP file from 96 to 97. Can someone tell me why ? Where is this number defined ?

  Here is my source. See if you can replicate it.

  #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>
  
  }
  
  
  
  
  
  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->oformat->audio_codec = AV_CODEC_ID_PCM_S16BE;
  
  
  
      /* 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;
  
      outCodecCtx = outAudioStream->codec;
  
      outCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
  
  
  
      /* select other audio parameters supported by the encoder */
  
      outCodecCtx->sample_rate = 8000;
  
      outCodecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
  
      outCodecCtx->channels = 2;
  
  
  
      /* open it */
  
      if (avcodec_open2(outCodecCtx, outCodec, NULL) &lt; 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      // PCM has no frame, so we have to explicitly specify
  
      outCodecCtx->frame_size = 1152;
  
  
  
      av_dump_format(outFormatCtx, 0, filename, 1);
  
  
  
      char buff[10000] = { 0 };
  
      ret = av_sdp_create(&amp;outFormatCtx, 1, buff, sizeof(buff));
  
      printf("%s", buff);
  
  
  
      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;
  
  
  
      /* 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; 400000; 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] = (uint16_t)(sin(t) * 10000);
  
  
  
              for (k = 1; k &lt; outCodecCtx->channels; k++)
  
                  samples[2 * j + k] = samples[2 * j];
  
              t += tincr;
  
          }
  
          t = (t > 50000) ? 0 : t;
  
  
  
          // 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) {
  
              write_frame(outFormatCtx, &amp;outCodecCtx->time_base, outAudioStream, &amp;pkt);
  
              av_packet_unref(&amp;pkt);
  
          }
  
  
  
          printf("i:%d\n", i); // waste some time to avoid over-filling jitter buffer
  
          printf("Audio: %d\t%d\n", samples[0], samples[1]); // waste some time to avoid over-filling jitter buffer
  
          printf("t: %f\n", t); // waste some time to avoid over-filling jitter buffer
  
      }
  
  
  
      /* 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) {
  
              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;
  
  
  
      av_register_all();
  
      avformat_network_init(); // for network streaming
  
  
  
      audio_encode_example("rtp://127.0.0.1:8554");
  
  
  
      return 0;
  
  }

  Update

  Curiously, running on Linux Ubuntu gives me the following instead :

  Output #0, rtp, to 'rtp://127.0.0.1:8554':
  
      Stream #0:0: Unknown: none (pcm_s16be)
  
  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 57.48.100
  
  m=application 8554 RTP/AVP 3

  Does anyone know why the stream has been changed from audio to application ?