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  • How to convert rtmp hevc video stream to srt av1 endpoint with ffmpeg ?

    20 juin 2024, par Lulík

    i want use ffmpeg to listen rtmp stream and send to srt endpoint.

    


    Flow : smartphone (camera) -> laptop (ffmpeg script) -> desktop (obs studio)

    


    ffmpeg script show warning message and in obs stuido i can see any video only audio.

    


    Thank you in advance.

    


    Console output while running script (error in the end is bcs i stoped sending data from phone) :

    


    ffmpeg version git-2024-06-20-8d6014d Copyright (c) 2000-2024 the FFmpeg developers
  built with gcc 12 (Debian 12.2.0-14)
  configuration: --enable-libsvtav1 --enable-libsrt
  libavutil      59. 24.100 / 59. 24.100
  libavcodec     61.  8.100 / 61.  8.100
  libavformat    61.  3.104 / 61.  3.104
  libavdevice    61.  2.100 / 61.  2.100
  libavfilter    10.  2.102 / 10.  2.102
  libswscale      8.  2.100 /  8.  2.100
  libswresample   5.  2.100 /  5.  2.100
Input #0, flv, from 'rtmp://192.168.0.194/s/streamKey':
  Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
  Stream #0:0: Video: hevc (Main), yuv420p(tv, smpte170m/bt470bg/smpte170m), 1080x1920, 10240 kb/s, 30 fps, 120 tbr, 1k tbn
  Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 131 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (hevc (native) -> av1 (libsvtav1))
  Stream #0:1 -> #0:1 (aac (native) -> mp2 (native))
Press [q] to stop, [?] for help
Svt[info]: -------------------------------------------
Svt[info]: SVT [version]:   SVT-AV1 Encoder Lib 595a874
Svt[info]: SVT [build]  :   GCC 12.2.0   64 bit
Svt[info]: LIB Build date: Jun 20 2024 14:25:08
Svt[info]: -------------------------------------------
Svt[info]: Number of logical cores available: 12
Svt[info]: Number of PPCS 76
Svt[info]: [asm level on system : up to avx2]
Svt[info]: [asm level selected : up to avx2]
Svt[info]: -------------------------------------------
Svt[info]: SVT [config]: main profile   tier (auto) level (auto)
Svt[info]: SVT [config]: width / height / fps numerator / fps denominator       : 1080 / 1920 / 120 / 1
Svt[info]: SVT [config]: bit-depth / color format                   : 8 / YUV420
Svt[info]: SVT [config]: preset / tune / pred struct                    : 10 / PSNR / random access
Svt[info]: SVT [config]: gop size / mini-gop size / key-frame type          : 641 / 16 / key frame
Svt[info]: SVT [config]: BRC mode / rate factor                     : CRF / 35 
Svt[info]: SVT [config]: AQ mode / variance boost                   : 2 / 0
Svt[info]: -------------------------------------------
Svt[warn]: Failed to set thread priority: Invalid argument
Output #0, mpegts, to 'srt://192.168.0.167:9998?mode=caller':
  Metadata:
    encoder         : Lavf61.3.104
  Stream #0:0: Video: av1, yuv420p(tv, smpte170m/bt470bg/smpte170m, progressive), 1080x1920, q=2-31, 120 fps, 90k tbn
      Metadata:
        encoder         : Lavc61.8.100 libsvtav1
  Stream #0:1: Audio: mp2, 44100 Hz, stereo, s16, 384 kb/s
      Metadata:
        encoder         : Lavc61.8.100 mp2
[mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading.
[in#0/flv @ 0x55ec9219cc40] Error during demuxing: Input/output error1990.7kbits/s speed=0.967x    
[out#0/mpegts @ 0x55ec922247c0] video:4431KiB audio:1138KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 6.374870%
frame=  723 fps= 31 q=35.0 Lsize=    5923KiB time=00:00:24.12 bitrate=2011.3kbits/s speed=1.04x


    


    I send video stream from mobile app over rtmp encoded with hevc to my laptop where running script ffmpeg -f flv -listen 1 -i rtmp://192.168.0.194/s/streamKey -c:v libsvtav1 -f mpegts srt://192.168.0.167:9998?mode=caller. On the desktop i have obs with media source input srt://192.168.0.167:9998?mode=listener.

    


    When i run ffmpeg script without video codec option (-c:v libsvtav1) its working fine and in obs i can see video from my phone camera. With the option i can not see video only audio.
I clearly dont understand warning message : [mpegts @ 0x55ec921d9540] Stream 0, codec av1, is muxed as a private data stream and may not be recognized upon reading..
Do I need specify codec (av1) in obs media source or my ffmpeg script is wrong ?

    


  • ffmpeg dash Segment offset

    18 mars 2019, par inkubux

    I’m trying to integrate live-transcoding like "plex" or "emby" with my application.

    I am able to serve dash content over to shaka-player or dash.js but only in ’live-mode’. But I want to enable seeking through the player.

    I looked at plex and to enable this they create their own mpd file with duration so the player will have a full seekbar.

    However when seeking the player will ask for a segment number eg : 449. I need to stop ffmpeg and restart with an offset (-ss &lt;<segment>>)</segment>, but ffmpeg will just restart a transcode session from segment 0 with an initial segment.

    What I want is to tell ffmpeg to start at a seekpoint but only output from segment number and now-on.

    When playing with hls and mpegts, I can tell ffmpeg to output at a certain segment : with the option -segment_start_number but this is not available for dash. And plex use their own transcoder based of ffmpeg with the option -skip_to_segment

    I tried to ’hack’ around by keeping a manual offset on my web-server, even if I serve the "supposed" right segment after the seek point dash.js and shaka-player can’t recover the stream.. VLC on the other habd is able to (probably more tolerent) to errors in segments.

    Is the supposed right segment after a seek in dash (contains the initial segment) or only the segment.

    Is ffmpeg able to start segmenting dash as a supposed segment (for seek and resume)

    The same technique works in hls with forced key frames and a custom m3u8 (with all the "predicted" segments) but calculating the right segment length and the right bandwidth is much harder and hackish and dash is more tolerant to variation.

    I would really like to be able to seek through my live transcoding video.

    For reference here is a custom mpd file I serve to enable "seeking" :

    <mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" type="static" suggestedpresentationdelay="PT1S" mediapresentationduration="PT49M2.920S" maxsegmentduration="PT2S" minbuffertime="PT10S">
       <period start="PT0S" duration="PT49M2.920S">
           <adaptationset segmentalignment="true">
               <segmenttemplate timescale="1" duration="1" initialization="$RepresentationID$/initial.mp4" media="$RepresentationID$/$Number$.m4s" startnumber="1">
               </segmenttemplate>
               <representation mimetype="video/mp4" codecs="avc1.640029" bandwidth="3766000" width="1920" height="1080">
               </representation>
           </adaptationset>
           <adaptationset segmentalignment="true">
               <segmenttemplate timescale="1" duration="1" initialization="$RepresentationID$/initial.mp4" media="$RepresentationID$/$Number$.m4s" startnumber="1">
               </segmenttemplate>
               <representation mimetype="audio/mp4" codecs="mp4a.40.2" bandwidth="188000" audiosamplingrate="48000">
                   <audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="6"></audiochannelconfiguration>
               </representation>
           </adaptationset>
       </period>
    </mpd>

    And here is the ffmpeg command to pull it off :

    ffmpeg -ss 0 -i movie.mkv -y -acodec aac -vcodec libx264 -f dash -min_seg_duration 1000000 -individual_header_trailer 0 -pix_fmt yuv420p -vf scale=trunc(min(max(iw\,ih*dar)\,1920)/2)*2:trunc(ow/dar/2)*2 -bsf:v h264_mp4toannexb -profile:v high -level 4.1 -map_chapters -1 -map_metadata -1 -preset veryfast -movflags frag_keyframe+empty_moov -use_template 1 -use_timeline 0 -remove_at_exit 1 -crf 23 -bufsize 7532k -maxrate 3766k -start_at_zero -threads 0 -force_key_frames expr:if(isnan(prev_forced_t),eq(t,t),gte(t,prev_forced_t+1)) -init_seg_name $RepresentationID$/0_initial.mp4 -media_seg_name $RepresentationID$/0_$Number$.m4s /transcoding_temp/Z1GVWEc/index.mpd

    The media_seg_name is where I prepend the custom seek_point let’s say I want to seek to segment 1233 the template would be :

    -media_seg_name $RepresentationID$/1233_$Number$.m4s

    and the segments would be 1233_1 1233_2 1233_* So I can serve the right segment after seek. but the player does not recover and still downloading subsequent segments. I guess since a new initial segment is generated and I somehow miss headers for continuous playback after seek but I’m probably wrong.

    Thanks for your help