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• pipe:0 : could not find codec parameters

  10 avril 2018, par Harish

  I am trying to store streaming video as a segments of 15 min,it is working fine with gst-launch-1.0 my command is

  gst-launch-1.0 -e --gst-debug-level=2 v4l2src device=/dev/video0 ! vspmfilter outbuf-alloc=true ! \
  
  video/x-raw,width=640,height=480,framerate=15/2 ! omxh264enc target-bitrate=10485760 num-p-frames=0 ! \
  
  mpegtsmux name="muxer" ! fdsink fd=1 \
  
  alsasrc device=hw:0,0 latency-time=30000 buffer-time=800000 ! audioconvert ! audio/x-raw,format=S16LE,channel=2,rate=44100 ! \
  
  faac bitrate=128000 ! muxer. \
  
  | /usr/bin/ffmpeg -v info -i pipe:0 -map 0 -c copy -f segment \
  
  -reset_timestamps 1 -segment_atclocktime 1 -segment_time 900 \
  
  -segment_list_type csv -segment_list_size 0 \
  
  -segment_list /home/root/ContinuousCapture/CamFiles.txt
  
  -segment_list_entry_prefix /home/root/ContinuousCapture/ \
  
  -segment_format mpegts -strftime 1 /home/root/ContinuousCapture/video_%g%m%d_%H%M%S.mpegts

  I changed same command to gstd,my command is

  gstd-client pipeline_create p v4l2src device=/dev/video0 ! ${textoverlay} text=${dev0} ! ${clockoverlay} ! vspmfilter outbuf-alloc=true ! \
  
  video/x-raw,width=640,height=480,framerate=15/2 ! omxh264enc target-bitrate=10485760 num-p-frames=0 \
  
  mpegtsmux name="muxer" ! fdsink fd=1 \
  
  alsasrc device=hw:0,0 latency-time=30000 buffer-time=800000 ! audioconvert ! audio/x-raw,format=S16LE,channel=2,rate=44100 ! \
  
  faac bitrate=128000 ! muxer. \ 
  
  gstd-client pipeline_play p | /usr/bin/ffmpeg  -v info -i pipe:0 -map 0 -c copy -f segment \
  
  -reset_timestamps 1 -segment_atclocktime 1 -segment_time 900 \
  
  -segment_list_type csv -segment_list_size 0 \
  
  -segment_list /home/root/ContinuousCapture/CamFiles.txt -segment_list_entry_prefix /home/root/ContinuousCapture/ \
  
  -segment_format mpegts -strftime 1 /home/root/ContinuousCapture/video_%g%m%d_%H%M%S.mpegts

  its giving me the following error

  libdir=/usr/lib --shlibdir=/usr/lib --datadir=/usr/share/ffmpeg --disable-mipsdsp --disable-mipsdspr2 --cpu=generic --pkg-config=pkg-config --enable-avcodec --enable-avdevice --enable-avfilter --enable-avformat --disable-avresample --enable-bzlib --enable-gpl --disable-libgsm --disable-indev=jack --disable-libvorbis --disable-libmp3lame --disable-openssl --enable-postproc --disable-libschroedinger --disable-sdl2 --disable-libspeex --enable-swresample --enable-swscale --enable-libtheora --disable-vaapi --disable-vdpau --disable-libvpx --enable-libx264 --enable-outdev=xv
  
  libavutil      55. 58.100 / 55. 58.100
  
  libavcodec     57. 89.100 / 57. 89.100
  
  libavformat    57. 71.100 / 57. 71.100
  
  libavdevice    57.  6.100 / 57.  6.100
  
  libavfilter     6. 82.100 /  6. 82.100
  
  libswscale      4.  6.100 /  4.  6.100
  
  libswresample   2.  7.100 /  2.  7.100
  
  libpostproc    54.  5.100 / 54.  5.100
  
  [mpegts @ 0x49450] Format mpegts detected only with low score of 2, misdetection possible!
  
  [mpegts @ 0x49450] Could not detect TS packet size, defaulting to non-FEC/DVHS
  
  pipe:0: could not find codec parameters

  Thank you

• Live555 truncates encoded data of FFMpeg

  22 novembre 2019, par Harshil Makwana

  I am trying to stream H264 based data using Live555 over RTSP.

  I am capturing data using V4L2, and then encodes it using FFMPEG and then passing data to Live555’s DeviceSource file, in that I using H264VideoStreamFramer class,

  Below is my codec settings to configure AVCodecContext of encoder,

  codec = avcodec_find_encoder_by_name(CODEC_NAME);
  
  if (!codec) {
  
      cerr << "Codec " << codec_name << " not found\n";
  
      exit(1);
  
  }
  
  
  
  c = avcodec_alloc_context3(codec);
  
  if (!c) {
  
      cerr << "Could not allocate video codec context\n";
  
      exit(1);
  
  }
  
  
  
  pkt = av_packet_alloc();
  
  if (!pkt)
  
      exit(1);
  
  
  
  /* put sample parameters */
  
  c->bit_rate = 400000;
  
  /* resolution must be a multiple of two */
  
  c->width = PIC_HEIGHT;
  
  c->height = PIC_WIDTH;
  
  /* frames per second */
  
  c->time_base = (AVRational){1, FPS};
  
  c->framerate = (AVRational){FPS, 1};
  
  c->gop_size = 10;
  
  c->max_b_frames = 1;
  
  c->pix_fmt = AV_PIX_FMT_YUV420P;
  
  c->rtp_payload_size = 30000;
  
  if (codec->id == AV_CODEC_ID_H264)
  
      av_opt_set(c->priv_data, "preset", "fast", 0);
  
  av_opt_set_int(c->priv_data, "slice-max-size", 30000, 0);
  
  /* open it */
  
  ret = avcodec_open2(c, codec, NULL);
  
  if (ret < 0) {
  
      cerr << "Could not open codec\n";
  
      exit(1);
  
  }

  And I am getting encoded data using avcodec_receive_packet() function. which will return AVPacket.

  And I am passing AVPacket’s data into DeviceSource file below is code snippet of my Live555 code :

  void DeviceSource::deliverFrame() {
  
      if (!isCurrentlyAwaitingData()) return; // we're not ready for the data yet
  
  
  
      u_int8_t* newFrameDataStart = (u_int8_t*) pkt->data;
  
      unsigned newFrameSize = pkt->size; //%%% TO BE WRITTEN %%%
  
      // Deliver the data here:
  
      if (newFrameSize > fMaxSize) { // Condition becomes true many times
  
          fFrameSize = fMaxSize;
  
          fNumTruncatedBytes = newFrameSize - fMaxSize;
  
      } else {
  
          fFrameSize = newFrameSize;
  
      }
  
      gettimeofday(&fPresentationTime, NULL); // If you have a more accurate time - e.g., from an encoder - then use that instead.
  
      // If the device is *not* a 'live source' (e.g., it comes instead from a file or buffer), then set "fDurationInMicroseconds" here.
  
      memmove(fTo, newFrameDataStart, fFrameSize);
  
  }

  But here, sometimes my packet’s size is getting more than fMaxSize value and as per LIVE555 logic it will truncate frame data, so that sometimes I am getting bad frames on my VLC,

  From Live555 forum, I get to know that encoder should not send packet whose size is more than fMaxSize value, so my question is :

  How to restrict encoder to limit size of packet ?

  Thanks in Advance,

  Harshil

• How to Configure nginx with stunnel to accept parameters for different FB Live Streams OR rtmps with FFMPEG

  15 septembre 2020, par Yogesh Agarwal

  I want to setup RTMPS and learned that only way around is by using nginx and stunnel. I have the setup and it works with just one configured key.

  



  I have several different keys and all are dynamic. A lot of different urls too.

  



  My Goal is to add a parameter or some way by which i can send the custom url to nginx and it can send to Stunnel, and it can read the custom url and forward the encrypted stream to that url.

  



  I am able to get everything done right via ngnix and stunnel but with preconfigured key only - But i want Dynamic key here.. just like a parameter where you can simply plug and send the stream.

  



  I even tried to set this way.

  



  push rtmp ://127.0.0.1:19350/rtmp/ ;

  



  so i can simple forward the stream to rtmp ://127.0.0.1:19350/rtmp/my-key and it takes my-key and forward the stream via stunnel to facebook. but i cannot get it work.

  



  I am about to bang my heads in walls. Kindly give me some pointers.. I am not sure how to do it via ffmpeg as it says it cannot find rtmps protocol.

  



  My Nginx Config :

  



  # RTMP configuration
rtmp {
    server {
        listen 1935; # Listen on standard RTMP port
        chunk_size 4000;
 # This application is to accept incoming stream
        application live {


                live on; # Allows live input from above
                exec_push rtmp://127.0.0.1:19350/rtmp/$name;
                allow play 127.0.0.1;
                dash on;
                dash_path /var/tmp/dashme;

                hls on; # Enable HTTP Live Streaming
                hls_cleanup on;
                hls_sync 100ms;
                hls_fragment 2s;
                hls_path /var/tmp/live/;


        }


  



  and My Stunnel Config :

  



  setuid = nobody
setgid = nobody
pid=/tmp/stunnel.pid
output = /var/log/stunnel.log
;include = /etc/stunnel/conf.d

[fb-live]
client = yes
accept = 127.0.0.1:19350
connect = live-api-s.facebook.com:443
;verifyChain = no