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• FFMPEG pushed RTMP stream not working on Android & iPhone

  1er décembre 2015, par BlackDivine

  I have to make a semi-live-stream. I used Nginx-rtmp module and then pushed content to it via ffmpeg using :

  ffmpeg -re -i content.mp4 -r 25 -f fvl "rtmp://rtmp.server.here"

  The stream runs fine when I open it in VLC from "rtmp ://rtmp.server.here"

  But I also have to make iPhone and Android apps that play these streams. And that’s the problem, the stream doesn’t work on Android and iPhone.

  If I use Wowza streaming cloud and stream to Wowza cloud instead of my own nginx-rtmp server then the same app written for Android & iPhone can playback the stream just fine.

  Now either nginx-rtmp is not working right, or what else ? I’ve also tried crtmpserver and the same thing happens.

  What I want to acheive :
  I have to develop a system where we can upstream a TV-Channel (have rights for it) to a server and then make a website, android app & iPhone app so consumers can watch the live channel.

  The uploading part I have a clue of, probably a TV tuner card and Open Broadcast Software to stream it to server. But the Live playback is new to me.

  ---

  UPDATE : I also used ffprobe and here’s the output. (See the last line)

  munir@munir-HP-ProBook-450-G2:~$ ffprobe rtmp://rtmp.server.here
  
  ffprobe version 2.6.2 Copyright (c) 2007-2015 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvo-aacenc --enable-libvidstab
  
    libavutil      54. 20.100 / 54. 20.100
  
    libavcodec     56. 26.100 / 56. 26.100
  
    libavformat    56. 25.101 / 56. 25.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.102 /  5. 11.102
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [flv @ 0x267cc60] Stream discovered after head already parsed
  
      Last message repeated 1 times
  
  Input #0, flv, from 'rtmp://stage.funworldpk.com/live':
  
    Metadata:
  
      Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
  
      displayWidth    : 320
  
      displayHeight   : 240
  
      fps             : 20
  
      profile         : 
  
      level           : 
  
    Duration: 00:00:00.00, start: 288.763000, bitrate: N/A
  
      Stream #0:0: Video: h264 (High), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 20 fps, 20 tbr, 1k tbn, 40 tbc
  
      Stream #0:1: Data: none
  
      Stream #0:2: Audio: aac (LC), 22050 Hz, stereo, fltp
  
  Unsupported codec with id 0 for input stream 1

  Update 2 :
  I got my stream working by using Licensed copy of Wowza streaming server. Everything works now. But obviously this will not be an option for everyone that’s why I am not posting it as an answer.

• FFmpeg starting manually but not with Systemd on boot

  23 juin 2021, par eKrajnak

  On Raspberry Pi 4 B 4GB with official Debian 10 image, I have /home/pi/run.sh script with following :

  


  #!/bin/bash
ffmpeg -nostdin -framerate 15 -video_size 1280x720 -input_format yuyv422  -i /dev/video0 -f alsa -i hw:Device \
    -af acompressor=threshold=-14dB:ratio=9:attack=10:release=1000 -c:a aac -ac 2 -ar 48000 -ab 160k \
    -c:v libx264 -pix_fmt yuv420p -b:v 3M -bf 1 -g 20 -flags +ilme+ildct -preset ultrafast \
    -streamid 0:0x101 -streamid 1:0x100 -mpegts_pmt_start_pid 4096 -mpegts_start_pid 0x259 -metadata:s:a:0 language="" -mpegts_service_id 131 -mpegts_transport_stream_id 9217 -metadata provider_name="Doesnt matter" -metadata service_name="Doesnt matter" \
    -minrate 3500 -maxrate 3500k -bufsize 4500k -muxrate 4000k  -f mpegts "udp://@239.1.67.13:1234?pkt_size=1316&bitrate=4000000&dscp=34" -loglevel debug < /dev/null > /tmp/ff3.log 2>&1


  


  Script is starting from console without problems. It takes audio from USB sound card and video from USB camera and creates UDP stream to IPTV. Then I created Systemd service :

  


  [Unit]
Description=Streamer
After=multi-user.target sound.target network.target

[Service]
ExecStart=/home/pi/run.sh
KillMode=control-group
Restart=on-failure
TimeoutSec=1

[Install]
WantedBy=multi-user.target
Alias=streaming.service


  


  After restarting Raspberry, script has started, but FFmpeg hangs on error failures in log :

  


  cur_dts is invalid st:0 (257) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:1 (256) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (257) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:1 (256) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (257) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:1 (256) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:0 (257) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)
cur_dts is invalid st:1 (256) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream)


  


  and will not start streaming to UDP target. But, if I manually login to SSH and issue systemctl stop streaming and then systemctl start streaming Ffmpeg starts successfully. What's different with service auto-start on boot ?

  


  Setting the "sleep timeout" at script begginging will not help. However, removing audio stream from FFmpeg config looks to solve auto-start on boot.

  


• Pass embedded closed captions with FFmpeg

  4 décembre 2020, par Andrew

  I am taking an HLS stream and outputting it through a DeckLink card to convert it to SDI. I need to pass the CEA-708 closed captions that are embedded in the stream, and it has to happen live. This is the command I'm currently using :

  



  ffmpeg.exe -loglevel info -re -i https://my-url.m3u8 -c:v wrapped_avframe -pix_fmt uyvy422 -vf scale=1280x720 -c:a pcm_s16le -ar 48000 -ac 2 -f decklink -s 1280x720 -r 60000/1001 -b:v 4M "DeckLink SDI (2)"

  



  I have tried adding the -teletext all command to no avail.

  



  Here is the output of FFprobe :

  



  Input #0, hls, from 'https://my-url.m3u8':
  Duration: N/A, start: 83023.890600, bitrate: N/A
  Program 0
    Metadata:
      variant_bitrate : 998000
    Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 640x360, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 998000
    Stream #0:1(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 998000
      comment         : Master Audio
    Stream #0:2: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 998000
  Program 1
    Metadata:
      variant_bitrate : 658000
    Stream #0:3: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 480x270, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 658000
    Stream #0:4(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 658000
      comment         : Master Audio
    Stream #0:5: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 658000
  Program 2
    Metadata:
      variant_bitrate : 1958000
    Stream #0:6: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 960x540, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 1958000
    Stream #0:7(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 1958000
      comment         : Master Audio
    Stream #0:8: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 1958000
  Program 3
    Metadata:
      variant_bitrate : 2658000
    Stream #0:9: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720, Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Metadata:
      variant_bitrate : 2658000
    Stream #0:10(eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 158 kb/s (default)
    Metadata:
      variant_bitrate : 2658000
      comment         : Master Audio
    Stream #0:11: Data: timed_id3 (ID3  / 0x20334449)
    Metadata:
      variant_bitrate : 2658000
Unsupported codec with id 100358 for input stream 2
Unsupported codec with id 100358 for input stream 5
Unsupported codec with id 100358 for input stream 8
Unsupported codec with id 100358 for input stream 11


  



  As you can see, closed captions are embedded in each of the VBR streams, but are getting stripped during the transcode to SDI. How can I pass these to view them ?