Sur d’autres sites (5304)

• How to convert RTSP stream into flv/swf Stream (w. ffmpeg) ?

  22 octobre 2012, par acy

  I want embed a webcam stream (From geovision video server) into a website. Unfortunately only the rtsp stream gives direct access to the video data.

  I tried a bunch of different variants. With this version I got no errors :

  openRTSP -b 50000 -w 352 -h 288 -f 5 -v -c -u admin password rtsp://xxxxxx.dyndns.org:8554/CH001.sdp | \
  
  ffmpeg -r 5 -b 256000 -f mp4 -i - http://127.0.0.1:8090/feed1.ffm

  Unfortunately I get no video. Sometimes I see a single frame of the webcam, but no livestream.

  This is my ffserver.conf

  Port 8090
  
  BindAddress 0.0.0.0
  
  MaxClients 200
  
  MaxBandwidth 20000
  
  CustomLog /var/log/flvserver/access.log
  
  
  
  NoDaemon
  
  
  
  # Server Status
  
  <stream>
  
  Format status
  
  </stream>
  
  
  
  <feed>
  
  File /tmp/feed1.ffm
  
  FileMaxSize 200K
  
  ACL allow 127.0.0.1
  
  </feed>
  
  
  
  # SWF output - great for testing
  
  <stream>
  
   # the source feed
  
   Feed feed1.ffm
  
   # the output stream format - SWF = flash
  
   Format swf
  
   #VideoCodec flv
  
   # this must match the ffmpeg -r argument
  
   VideoFrameRate 5
  
   # another quality tweak
  
   VideoBitRate 256K
  
   # quality ranges - 1-31 (1 = best, 31 = worst)
  
   VideoQMin 1
  
   VideoQMax 3
  
   VideoSize 352x288
  
   # wecams don&#39;t have audio
  
   NoAudio
  
  </stream>

  What am I doing wrong ? THe test.swf seems to load forever...

• Why does use of H264 in sender/receiver pipelines introduce just HUGE delay ?

  24 janvier 2012, par Serguey Zefirov

  When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine ! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong.

  I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex :

  This is sender pipeline :
  # !/bin/sh

  gst-launch -v gstrtpbin name=rtpbin \
  
          v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
  
                    rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000                            \
  
                    rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false    \
  
                    udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           \
  
          pulsesrc ! audioconvert ! audioresample  ! audio/x-raw-int,rate=16000 !    \
  
                    speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1                   \
  
                    rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002                            \
  
                    rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false    \
  
                    udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

  Receiver pipeline :

   !/bin/sh

  gst-launch -v\
  
      gstrtpbin name=rtpbin                                          \
  
      udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \
  
              port=5000 ! rtpbin.recv_rtp_sink_0                                \
  
          rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink                    \
  
       udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               \
  
       rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        \
  
      udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \
  
              port=5002 ! rtpbin.recv_rtp_sink_1                                \
  
          rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \
  
       udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               \
  
       rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false

  Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time.

  Then I changed pipelines to use H264 along the video path.

  The sender becomes :
  # !/bin/sh

  gst-launch -v gstrtpbin name=rtpbin \
  
          v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \
  
                    rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000                            \
  
                    rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false    \
  
                    udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           \
  
          pulsesrc ! audioconvert ! audioresample  ! audio/x-raw-int,rate=16000 !    \
  
                    speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1                   \
  
                    rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002                            \
  
                    rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false    \
  
                    udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

  And receiver becomes :
  # !/bin/sh

  gst-launch -v\
  
      gstrtpbin name=rtpbin                                          \
  
      udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \
  
              port=5000 ! rtpbin.recv_rtp_sink_0                                \
  
          rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink                    \
  
       udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               \
  
       rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        \
  
      udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \
  
              port=5002 ! rtpbin.recv_rtp_sink_1                                \
  
          rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \
  
       udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               \
  
       rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false

  This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

• Changing int main() to JNI interface prototype

  13 mars 2012, par iSun

  I changed ffmpeg.c according following link :

  http://www.roman10.net/how-to-port-ffmpeg-the-program-to-androidideas-and-thoughts/

  He said the change main () to JNI interface prototype. Well, I'm not familiar with JNI interface prototype, but I read an article about JNI and change it accordingly.

  Can anyone look at my code to see is this true or not ?

  JNIEXPORT jint JNICALL Java_com_ffmpegtest_MainActivity_main(JNIEnv *pEnv, int argc, char **argv) {
  
  int64_t ti;
  
  
  
  av_log_set_flags(AV_LOG_SKIP_REPEATED);
  
  
  
  if(argc>1 && !strcmp(argv[1], "-d")){
  
  run_as_daemon=1;
  
  verbose=-1;
  
  av_log_set_callback(log_callback_null);
  
  argc--;
  
  argv++;
  
  
  
  }
  
  
  
  avcodec_register_all();
  
  #if CONFIG_AVDEVICE
  
  avdevice_register_all();
  
  #endif
  
  #if CONFIG_AVFILTER
  
  avfilter_register_all();
  
  #endif
  
  av_register_all();
  
  
  
  #if HAVE_ISATTY
  
  if(isatty(STDIN_FILENO))
  
  avio_set_interrupt_cb(decode_interrupt_cb);
  
  #endif
  
  
  
  init_opts();
  
  
  
  if(verbose>=0)
  
  show_banner();
  
  
  
  /* parse options */
  
  parse_options(argc, argv, options, opt_output_file);
  
  
  
  if(nb_output_files <= 0 && nb_input_files == 0) {
  
  show_usage();
  
  fprintf(stderr, "Use -h to get full help or, even better, run 'man ffmpeg'\n");
  
  ffmpeg_exit(1);
  
  }
  
  
  
  /* file converter / grab */
  
  if (nb_output_files <= 0) {
  
  fprintf(stderr, "At least one output file must be specified\n");
  
  ffmpeg_exit(1);
  
  }
  
  
  
  if (nb_input_files == 0) {
  
  fprintf(stderr, "At least one input file must be specified\n");
  
  ffmpeg_exit(1);
  
  }
  
  
  
  ti = getutime();
  
  if (transcode(output_files, nb_output_files, input_files, nb_input_files,
  
  stream_maps, nb_stream_maps) < 0)
  
  ffmpeg_exit(1);
  
  ti = getutime() - ti;
  
  if (do_benchmark) {
  
  int maxrss = getmaxrss() / 1024;
  
  printf("bench: utime=%0.3fs maxrss=%ikB\n", ti / 1000000.0, maxrss);
  
  }
  
  
  
  return ffmpeg_exit(0);
  
  }