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• ffmpeg -Video live feed

  22 juin 2019, par Rajeev

  I am using AMCREST security camera at my home. My objective is to get a live feed one of IP camera attached to my NVR to a webportal using rtsp ://My environment is Raspberry pi.

  I am able to successfully start the ffserver but conversion is failing when I am trying pass the input video and stream it to video.ffm

  I have tried various combination of parameters in the command but the below one seems to be very close where I got only one error ( av_interleaved_write_frame() : Connection reset by peer)

  $ffmpeg  -thread_queue_size 800  -i "rtsp://home:Home1234@192.168.1.32:554/cam/realmonitor?channel=4&subtype=0" -f lavfi -i aevalsrc=0  http://127.0.0.1:8090/video.ffm 

  ******* ffmpeg server configuration file content : etc/livestream.conf

  #Default port
  
  HTTPPort 8090
  
  HTTPBindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 100000
  
  CustomLog -
  
  
  
  #############################################################
  
  
  
  <feed>
  
    File /tmp/video.ffm                    # this creates a temp video.ffm file where streams are read/write
  
    FileMaxSize 0.5G
  
    ACL allow localhost
  
    ACL allow 127.0.0.1
  
    ACL allow 192.168.0.0 192.168.255.255
  
  </feed>
  
  
  
  <stream stream="stream">
  
  # streaming for webm file
  
  # run : ffserver -f /etc/ffserver.conf
  
  # run : ffmpeg -i videoname.mp4 http://localhost:8090/video.ffm
  
  # error : encoder setup failed
  
    Feed video.ffm
  
    Format webm
  
  
  
  # Audio settings
  
    AudioCodec vorbis
  
    AudioBitRate 64                             # Audio bitrate
  
  
  
  # Video settings
  
    VideoCodec libvpx
  
    VideoSize 720x486                           # Video resolution
  
    VideoFrameRate 30                           # Video FPS
  
    AVOptionVideo flags +global_header          # Parameters passed to encoder
  
  
  
    AVOptionVideo cpu-used 0
  
    AVOptionVideo qmin 10                       # lower the better, min 0
  
    AVOptionVideo qmax 42                       # higher outputs bad quality, max 63
  
    AVOptionVideo quality good
  
    AVOptionAudio flags +global_header
  
    PreRoll 15
  
    StartSendOnKey
  
    VideoBitRate 400                            # Video bitrate
  
  </stream>
  
  
  
  
  
  
  
  ###########################################################################
  
  
  
  # Audio only
  
  # run ffmpeg -i audio.mp3 http://localhost:8090/audio.ffm
  
  # run http://localhost:8090/audio in vlc or browser
  
  
  
  <feed>
  
    File /tmp/audio.ffm
  
    FileMaxSize 1G
  
    ACL allow localhost
  
    ACL allow 127.0.0.1
  
    ACL allow 192.168.0.0 192.168.255.255
  
  </feed>
  
  
  
  <stream audio="audio">
  
    Feed audio.ffm
  
    Format mp2                                     #audio format
  
    AudioCodec libmp3lame                          #audio codec
  
    AudioBitRate 64                                #audio bitrate
  
    AudioChannels 1                                #audio channel, 1 for mono and 2 for stereo
  
    AudioSampleRate 44100
  
    NoVideo                                        #discard video
  
  </stream>
  
  
  
  ####################################################################
  
  #view status of ffserver
  
  <stream>
  
    Format status
  
    ACL allow localhost
  
    ACL allow 192.168.0.0 192.168.255.255
  
  </stream>
  
  
  
  
  
  
  
  # Redirect index.html to the appropriate site
  
  
  
  <redirect>
  
    URL http://www.ffmpeg.org/
  
  </redirect>

  *****output of ffmpeg server running successfully on separate console***

  $ ffserver -f /etc/livestream.conf

  ffserver version 3.2.14-1 deb9u1+rpt1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
  configuration : —prefix=/usr —extra-version=’1 deb9u1+rpt1’ —toolchain=hardened —libdir=/usr/lib/arm-linux-gnueabihf —incdir=/usr/include/arm-linux-gnueabihf —enable-gpl —disable-stripping —enable-avresample —enable-avisynth —enable-gnutls —enable-ladspa —enable-libass —enable-libbluray —enable-libbs2b —enable-libcaca —enable-libcdio —enable-libebur128 —enable-libflite —enable-libfontconfig —enable-libfreetype —enable-libfribidi —enable-libgme —enable-libgsm —enable-libmp3lame —enable-libopenjpeg —enable-libopenmpt —enable-libopus —enable-libpulse —enable-librubberband —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-libzmq —enable-libzvbi —enable-omx —enable-omx-rpi —enable-mmal —enable-openal —enable-opengl —enable-sdl2 —enable-libdc1394 —enable-libiec61883 —arch=armhf —enable-chromaprint —enable-frei0r —enable-libopencv —enable-libx264 —enable-shared
  libavutil 55. 34.101 / 55. 34.101
  libavcodec 57. 64.101 / 57. 64.101
  libavformat 57. 56.101 / 57. 56.101
  libavdevice 57. 1.100 / 57. 1.100
  libavfilter 6. 65.100 / 6. 65.100
  libavresample 3. 1. 0 / 3. 1. 0
  libswscale 4. 2.100 / 4. 2.100
  libswresample 2. 3.100 / 2. 3.100
  libpostproc 54. 1.100 / 54. 1.100
  /etc/livestream.conf:45 : Setting default value for audio sample rate = 22050. Use NoDefaults to disable it.
  /etc/livestream.conf:45 : Setting default value for audio channel count = 1. Use NoDefaults to disable it.
  /etc/livestream.conf:45 : Setting default value for video bit rate tolerance = 100000. Use NoDefaults to disable it.
  /etc/livestream.conf:45 : Setting default value for video rate control equation = tex^qComp. Use NoDefaults to disable it.
  /etc/livestream.conf:45 : Setting default value for video max rate = 13749264. Use NoDefaults to disable it.
  /etc/livestream.conf:45 : Setting default value for video buffer size = 800000. Use NoDefaults to disable it.
  Fri Jun 21 19:43:59 2019 FFserver started.

  ---

  2nd Console

  $$ffmpeg  -thread_queue_size 800  -i "rtsp://home:Home1234@192.168.1.32:554/cam/realmonitor?channel=4&subtype=0" -f lavfi -i aevalsrc=0  http://127.0.0.1:8090/video.ffm 

  *************************Output**********

  Input #0, rtsp, from 'rtsp://home:Home1234@192.168.1.32:554/cam/realmonitor?channel=4&subtype=0':
  
    Metadata:
  
      title           : Media Server
  
    Duration: N/A, start: 0.290000, bitrate: N/A
  
      Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709, progressive), 2304x1296 [SAR 1:1 DAR 16:9], 20 fps, 250 tbr, 90k tbn, 40 tbc
  
  Input #1, lavfi, from 'aevalsrc=0':
  
    Duration: N/A, start: 0.000000, bitrate: 2822 kb/s
  
      Stream #1:0: Audio: pcm_f64le, 44100 Hz, mono, dbl, 2822 kb/s
  
  [swscaler @ 0x256dd80] deprecated pixel format used, make sure you did set range correctly
  
  [libvpx @ 0x2564190] v1.6.1
  
  Output #0, ffm, to 'http://127.0.0.1:8090/video.ffm':
  
    Metadata:
  
      title           : Media Server
  
      creation_time   : now
  
      encoder         : Lavf57.56.101
  
      Stream #0:0: Audio: vorbis (libvorbis), 22050 Hz, mono, fltp, 64 kb/s
  
      Metadata:
  
        encoder         : Lavc57.64.101 libvorbis
  
      Stream #0:1: Video: vp8 (libvpx), yuv420p, 720x486 [SAR 6:5 DAR 16:9], q=10-42, 400 kb/s, 20 fps, 1000k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc57.64.101 libvpx
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/0 buffer size: 800000 vbv_delay: -1
  
  Stream mapping:
  
    Stream #1:0 -> #0:0 (pcm_f64le (native) -> vorbis (libvorbis))
  
    Stream #0:0 -> #0:1 (h264 (native) -> vp8 (libvpx))
  
  Press [q] to stop, [?] for help
  
  frame=    2 fps=1.2 q=0.0 size=       8kB time=00:00:00.03 bitrate=1966.0kbits/s dup=1 drop=0 speed=0.0av_interleaved_write_frame(): Connection reset by peer
  
  Error writing trailer of http://127.0.0.1:8090/video.ffm: Connection reset by peer frame=    2 fps=1.1 q=0.0 Lsize=      40kB time=00:00:00.03 bitrate=9830.2kbits/s dup=1 drop=0 speed=0.0189x    
  
  video:29kB audio:0kB subtitle:0kB other streams:0kB global headers:4kB muxing overhead: 36.183796%
  
  Conversion failed

  My expectation is that ffmpeg will start writing data to video.ffm located in /tmp directory so that I can read the data from our browser or vlc media player by just entering the following link
  http://localhost:8090/stream

  ****** Update after 2 hr******

  I made slight change to the command parameter and my out changed as well.What it looks like the temporary video file that got generated in /tmp folder is not getting consumes by video.ffm ( I may be wrong in my analysis)

  ffmpeg  -thread_queue_size 1200  -i "rtsp://home:Home1234@192.168.1.32:554/cam/realmonitor?channel=4&subtype=0" -f lavfi -i aevalsrc=0 -override_ffserver  http://127.0.0.1:8090/video.ffm

  ***************** Output ****

  Past duration 0.660332 too large     376kB time=00:00:01.03 bitrate=2978.9kbits/s dup=3 drop=5 speed=0.171x    
  
  Past duration 0.637352 too large     840kB time=00:00:03.72 bitrate=1847.5kbits/s dup=14 drop=5 speed=0.213x    
  
  Past duration 0.678307 too large
  
  Past duration 0.713280 too large    1180kB time=00:00:05.52 bitrate=1749.0kbits/s dup=21 drop=5 speed=0.218x    
  
  Past duration 0.901085 too large    1372kB time=00:00:06.72 bitrate=1670.4kbits/s dup=26 drop=5 speed=0.22x     
  
  Past duration 0.948051 too large    2456kB time=00:00:12.97 bitrate=1551.1kbits/s dup=51 drop=5 speed=0.226x    
  
  [rtsp @ 0x1fd7670] Thread message queue blocking; consider raising the thread_queue_size option (current value: 1200)
  
  Past duration 0.713280 too large    3336kB time=00:00:17.64 bitrate=1549.0kbits/s dup=70 drop=5 speed=0.228x    
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1537.0kbits/s dup=76 drop=5 speed=0.229x    
  
  [rtsp @ 0x1fd7670] RTP: missed 30 packets
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1533.3kbits/s dup=78 drop=5 speed=0.228x    
  
  [rtsp @ 0x1fd7670] RTP: missed 134 packets
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1539.5kbits/s dup=82 drop=5 speed=0.229x    
  
  [rtsp @ 0x1fd7670] RTP: missed 111 packets
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1526.3kbits/s dup=84 drop=5 speed=0.229x    
  
  [rtsp @ 0x1fd7670] RTP: missed 19 packets
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1521.7kbits/s dup=92 drop=5 speed=0.23x     
  
  [rtsp @ 0x1fd7670] RTP: missed 626 packets
  
  Past duration 0.651329 too large    4408kB time=00:00:23.64 bitrate=1527.0kbits/s dup=94 drop=5 speed=0.23x    
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1516.7kbits/s dup=94 drop=5 speed=0.23x    
  
  [rtsp @ 0x1fd7670] RTP: missed 134 packets
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1521.3kbits/s dup=98 drop=5 speed=0.231x    
  
  [rtsp @ 0x1fd7670] RTP: missed 123 packets
  
  Past duration 0.633354 too large    4608kB time=00:00:24.98 bitrate=1511.0kbits/s dup=99 drop=5 speed=0.23x     
  
  [rtsp @ 0x1fd7670] max delay reached. need to consume packetbitrate=1523.4kbits/s dup=99 drop=5 speed=0.231x    

• Intermittent segfault with docker no-new-privileges

  19 juin 2019, par whadhack

  Anyone experienced intermittent segfault when —security-op=no-new-privileges has been turned on ? Is there a way to prevent that , short of privileged mode ?

  Docker container with "—security-op=no-new-privileges" has been enabled.

  avcodec_send_packet()

  No segfault as in privileged mode is the expected result.

• wowza multiple files in one playlist

  14 février 2014, par kalafun

  Is it possible to create a stream in wowza from multiple files ? So these file would be played in a row after each other ? As far as I know, I can only stream from one file being in the content directory..

  1.) I would like to split that one file for my own reasons, to add some security to it etc... , and then to create the playlist from these multiple files and publish it for streaming.. so it won't take that much time comparing to the second way.

  2.) Or do I need to put these multiple files back together and then publish the playlist ?

  I would also like to consider the time of the playlist being created even using a big file. I am using ffmpeg to split the file into smaller pieces using a script.

  Therefore it would be automatic, when a user would request a stream, I run the script that splits the files and creates the playlist for user..

  I hope I didn't take it from the wrong way. Help please