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• Live streaming Rmtp node-media-server real server not creating video files but wont fail (OBS)

  3 octobre 2020, par Orgil

  I have made a live streaming server with node-media-server and it works well in local but when i put it on a real server (ubuntu, nginx) its just creating the folder but not creating video files and no errors given OBS the streaming sofware not failing. is it something to do with ffmpeg ? maybe needed packages didnt install or ? idk whats going on :D

  


  Cofig of node-media-server :

  


  rtmp_server: {
    rtmp: {
        port: 1935,
        chunk_size: 60000,
        gop_cache: false,
        ping: 60,
        ping_timeout: 30
    },
    http: {
        port: 8088,
        mediaroot: '/home/ubuntu/projects/amjilt_media/media',
        allow_origin: '*'
    },
    trans: {
        ffmpeg: '/usr/bin/ffmpeg',
        // ffmpeg: 'C:/Users/User/Desktop/ffmpeg/bin/ffmpeg.exe',
        tasks: [
            {
                app: 'live',
                hls: true,
                hlsFlags: '[hls_time=2:hls_list_size=3:hls_flags=delete_segments]',
                dash: true,
                dashFlags: '[f=dash:window_size=3:extra_window_size=5]'
            }
        ]
    }
}


  


• HEVC/H.265 interlaced format support in ffmpeg or VLC

  30 décembre 2020, par Ernestas Gruodis

  "Music Box Russia" channel over satellite transmits in HEVC 1920x1080 25fps interlaced - and after recording VLC recognizes file as 50 fps, and resolution 1920x540 - half a height. But on satellite tuner the player works fine - it plays a file as 1920x1080 25fps... When we can expect support for HEVC/H.265 interlaced ? Here is recorded file (Garry Grey & Eva Miller - wtf). Also - a lot of lost frames in VLC player statistics..

  


  EDIT :

  


  I found some interesting info how in HEVC the interlace video content can be indicated here :

  


  


  Unlike to H.264/AVC, interlace-dedicated coding in HEVC is not exist :

  


  

  • No mixed frame-field interaction (like PAFF in H.264/AVC)

  


  • No interlace scanning of transform coefficients

  


  • No correction MVX[1] (or y-component of MV) if current and reference pictures are in different polarity (top-bottom or
bottom-top).

  


  


  However, in HEVC the interlace video content can be indicated
(signaled in VPS/SPS and pic_timing SEI messages the latter are
transmitted for every picture in the sequence). Interlace-related
setting :

  


  


  • in VPS/SPS set general_interlaced_source_flag=1 and general_progressive_source_flag=0. Indeed, the HEVC standard says :

    


    if general_progressive_source_flag is equal to 0 and
general_interlaced_source_flag is equal to 1, the source scan type of
the pictures in the CVS should be interpreted as interlaced only.

    


  


  • in VPS/SPS set general_frame_only_constraint_flag=0

    


  


  • in SPS VUI set field_seq_flag=1 and frame_field_info_present_flag=1. Notice that if these flags are ON
then picture timing SEIs shall be present for each picture.

    


  


  • transmission of Picture Timing SEI per picture with the following parameters :

    


    source_scan_type = 0 to indicate interlace mode
for top field picture signal pict_struct=1 and for bottom field picture pict_struct=2

    


  


  


  


  Perhaps it is possible to pass these parameters to ffmpeg/vlc before playing a file ?

  


• Convert mp3 to AAC with mpeg-2 container (FFMPEG)

  18 mars 2016, par jsurf

  I’m trying to convert an mp3 audio file to an AAC file with FFMPEG, and I need the audio to be wrapped in an MPEG-2 container.
  The resulting AAC file needs to be AAC-LC (Low Complexity), 1-channel, CBR mode, 44100 sample rate, and 48kb/s bitrate, so I use this command :

  ffmpeg -y -i input.mp3 -ar 44100 -ab 48k -acodec libfdk_aac -ac 1 output.aac

  But when I examine the ADTS headers, the audio file is always being wrapped in an MPEG-4 container. I have tried all the codecs listed here but I still end up with an mpeg-4 container wrapped around the audio : http://trac.ffmpeg.org/wiki/AACEncodingGuide.

  Here are the headers I get when examining the AAC output file :

  mpeg_type : ’MPEG4’,
  profile : 2,
  profile_name : ’AAC LC’,
  sample_freq : 44100,
  channel_config : 1,
  channels : 1,
  frame_length : 139,
  buffer_fullness : 157,
  number_of_frames : 1,
  frames_per_sec : 43.06640625

  Any ideas as to why ffmpeg wraps an mp4 container around the audio ? Can I get around this somehow ? Are there any other encoders I can try aside from FFMPEG ? I was giving FAAC encoder a shot and it gives me the proper encoding and ADTS headers, but alas it does not support mp3, only WAV.