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• FFMPEG when live streaming sends a message and exits after some frames were sent

  30 octobre 2020, par jstuardo

  when doing an streaming with FFMPEG all works perfectly until I get these messages and then, ffmpeg.exe exits :

  


  av_interleaved_write_frame(): Unknown error

frame= 1224 fps=3.4 q=13.0 size=    2758kB time=00:01:21.94 bitrate= 275.8kbits/s speed=0.226x    

av_interleaved_write_frame(): Unknown error

[flv @ 000001e310e8a1c0] Failed to update header with correct duration.

[flv @ 000001e310e8a1c0] Failed to update header with correct filesize.

Error writing trailer of rtmp://example.com/s/2b32abdc-130c-43e5-997e-079e69d1fd7f: Error number -10053 occurred

frame= 1224 fps=3.4 q=13.0 Lsize=    2758kB time=00:01:21.98 bitrate= 275.6kbits/s speed=0.226x    

video:2481kB audio:221kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.084671%

[libx264 @ 000001e310ad6080] frame I:41    Avg QP:10.29  size: 57664

[libx264 @ 000001e310ad6080] frame P:1183  Avg QP:13.52  size:   148

[libx264 @ 000001e310ad6080] mb I  I16..4: 100.0%  0.0%  0.0%

[libx264 @ 000001e310ad6080] mb P  I16..4:  0.1%  0.0%  0.0%  P16..4:  0.2%  0.0%  0.0%  0.0%  0.0%    skip:99.7%

[libx264 @ 000001e310ad6080] coded y,uvDC,uvAC intra: 10.9% 7.1% 5.4% inter: 0.0% 0.1% 0.0%

[libx264 @ 000001e310ad6080] i16 v,h,dc,p: 84%  6%  6%  4%

[libx264 @ 000001e310ad6080] i8c dc,h,v,p: 91%  6%  3%  1%

[libx264 @ 000001e310ad6080] kb/s:248.98

[aac @ 000001e310a46d40] Qavg: 108.454

Conversion failed!


  


  Normally, the messages I received are similar to this :

  


  frame= 1196 fps=3.4 q=13.0 size=    2692kB time=00:01:20.08 bitrate= 275.4kbits/s speed=0.227x    


  


  Which are the expected messages. Sometimes, I received this message, but this does not cause ffmpeg.exe to exit :

  


  Input #0, matroska,webm, from 'pipe:':

  Metadata:

    encoder         : Chrome

  Duration: N/A, start: 0.000000, bitrate: N/A

    Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)

    Stream #0:1(eng): Video: h264 (Constrained Baseline), yuv420p(progressive), 1920x1080, SAR 1:1 DAR 16:9, 30.30 fps, 14.99 tbr, 1k tbn, 60 tbc (default)


  


  What may be happening ? maybe it is a problem of the RTMP server ? or something is wrong with FFMPEG ?

  


  This version of FFMPEG.EXE is for windows. The programming language is C# from where I am launching FFMPEG.EXE process.

  


  As I told, this happens after several frames sent to the server. Only once, this problem occured after a few frames sent. That is why I suspect that the RTMP server is the problem.

  


  EDIT : This is the command :

  


  FFMPEG -i - -c:v libx264 -preset ultrafast -tune zerolatency -max_muxing_queue_size 1000 -bufsize 5000 -r 15 -g 30 -keyint_min 30 -x264opts keyint=30 -crf 25 -pix_fmt yuv420p -profile:v baseline -level 3 -c:a aac -b:a 22k -ar 22050 -f flv rtmp://rtmp.xxxx.yyyy


  


  Regards
Jaime

  


• How to restream multicast stream with ffmpeg

  26 octobre 2020, par verb

  I am new to ffmpeg and need to restream multicast and scale it. Tried different parameters and i have managed to restream and scale but it always appear some pat,pmt or pcr error and som interuptions in the stream appear.The input stream is cbr 14Mbit and i try to set the bitrate as 6Mbit please check my config and if you notice something wrong let me know :

  


  


  ffmpeg -re -i "udp ://@238.252.250.9:5000 ?overrun_nonfatal=1&fifo_size=1000000&bitrate=70000000&pkt_size=188" -map 0:0 -map 0:2 -b:v 3000k -minrate 3000k -maxrate 4000k -bufsize 8000K -pcr_period 20 -flush_packets 0 -tune zerolatency -preset ultrafast -threads 2 -c:a copy -qmax 12 -f mpegts -muxrate 6M "udp ://@239.253.251.13:5505 ?pkt_size=188&overrun_nonfatal=1&localaddr=10.253.251.66&bitrate=6000000"

  


  


  here is the input stream :

  


  Input #0, mpegts, from 'udp://@238.252.250.9:5000':
  Duration: N/A, start: 46612.831967, bitrate: N/A
  Program 2002 
    Metadata:
      service_name    : RT Doc HD
      service_provider: GLOBECAST
    Stream #0:0[0x7e5]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc
    Stream #0:1[0x7e6](eng): Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, stereo, fltp, 192 kb/s
    Stream #0:2[0x7e7](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 192 kb/s


  


  I don't understand all parameters especially the parameters concerning input/output udp stream so please help me to solve the correct command.

  


• ffmpeg rtp-stream with gsm-codec

  15 octobre 2020, par Birgit

  I want to use ffmpeg for encoding and decoding gsm. I built ffmpeg with the --enable-libgsm option.

  


  I can now use the ffmpeg-command-line-tool to read gsm-encoded files, convert files to gsm, and also receive a gsm-encoded rtp stream.
So therefore I think the gsm-encoder and gsm-decoder are working properly.

  


  But for some reason I am not able to send and gsm-encoded rtp-stream.

  


  I tried the following comands :

  


  ffmpeg -re -i test.wav -c:a libgsm -f rtp rtp://127.0.0.1:5000

  


  ffmpeg -re -i test.wav -c:a gsm -f rtp rtp://127.0.0.1:5000

  


  I receive the error : Unsupported codec gsm. Could not write header for output file.

  


  I tried to use gdb to see what's going on. I think the problem is that in the file libavformat/rtpenc.c:49 gsm is not under the supported codecs. Does that mean it is not possible to use ffmpeg to create a gsm-encoded rtp-stream ? Is there a workaround, to overcome this issue ?

  


  I would appreciate any help and hints what I could try. :)