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• How to Stream With FFmpeg and NGINX RTMP

  2 octobre 2023, par willowen100

  I'm trying to stream from OBS (open broadcast software) on my Windows PC to NGINX+RTMP also installed on the same PC. I have set a bitrate of 20,000Kbps in OBS which will be the foundation bitrate for the multiple streams I aim to setup within NGINX.

  



  I would like to be able to stream into NGINX and then on-the-fly use FFmpeg to transcode the stream to comply with the streaming site I intend to broadcast to, for example Twitch.tv.

  



  I can view my stream via VLC if I use the network path rtmp ://localhost/live/test. However, when I'm on Twitch's inspector site to see if my stream is coming thorugh, I'm not receiving anything. I have no idea if my FFmpeg is working or there is something wrong with my NGINX configuration below.

  



  If someone could shed some light of where I might be going wrong please that would be greatly appreciated.

  



  nginx.conf

  



  #user www-data;
worker_processes  1;

events {
    worker_connections  1024;
}

http { 
    server_tokens off;

    include mime.types;
    default_type application/octet-stream;
    sendfile off;
    keepalive_timeout 65;

    server {
        listen 80;
        server_name localhost;

        # make a internal server page and put it in html
        error_page 500 502 503 504 /50x.html;
        location = /50x.html {
            root html;
        }
    }
}

rtmp {
    server {
        listen 1935;
        chunk_size 8192;

        application live {
            live on;
            #interleave on;
            #wait_video on;
            record off;

            # Twitch
            exec_push "D:\Users\Will\Downloads\ffmpeg\bin"
                -i rtmp://localhost/source/$name 
                -c:v libx264 
                -c:a copy 
                -preset veryfast 
                -profile:v high 
                -level 4.1
                -x264-params "nal-hrd=cbr" "opencl=true"
                -b:v 8000K 
                -minrate 8000K 
                -maxrate 8000K
                -keyint 2
                -s 1920x1080
                push rtmp://live-lhr03.twitch.tv/app/STREAM_KEY;
        }
    }
}


  



  Many thanks

  



  UPDATE 1

  



  For the sake of simplicity I'm testing OBS, NGINX and FFmpeg all on the same physical computer, a Windows PC. Once everything is working I will port NGINX and FFmpeg to my Linux PC.

  



  I'm using a pre-compiled version of NGINX with the RTMP module baked in. I've also downloaded the latest FFmpeg libraries which I have set a path environment variable for in Windows so that FFmpeg commands can be called in CommandPrompt/PowerShell.

  



  Here's the path I'm trying to take :-

  



  OBS is encoding x264 at 20,000Kbps and it's destination is a RTMP application in NGINX called 'live'. From here I want to encode the one stream derived from OBS into several smaller bandwidth streams so that I can comply with streaming service's requirements such as Twitch and Mixer for example.

  



  At the end of the FFmpeg parameters do I push the output directly to Twitch or take the output of FFmpeg and send back into a second RTMP application on NGINX and then push out to Twitch ?

  



  One advantage of pushing FFmpeg's output back into NGINX before going off to the external stream service is I can open the FFmpeg transcoded stream through a RTMP supported player such as VLC for example, allowing me to view the compressed output.

  



  Another question I have is, can the FFmpeg parameters be put on separate lines or do they have to all in one line ?

  



  This is a really good site I have been referring back to

  



  https://blog.twitch.tv/en/2017/10/10/live-video-transmuxing-transcoding-f-fmpeg-vs-twitch-transcoder-part-i-489c1c125f28/

  


• ffplay startup time proportional to specified framerate

  17 décembre 2019, par bremen_matt

  I am playing a video over http using ffplay. The call I am using looks like this :

  ffplay -framerate 30 -fflags nobuffer -flags low_delay -autoexit -i http://localhost:8880

  The video is an H.264 encoding where (my understanding is a bit unclear here) it is something like a "raw" H.264 stream without timestamps.

  My primary concern is to get video displayed with low latency. In that regard, the video is fine.

  The issue is with the framerate and with the startup time.

  The video source is emitting frames as soon as they are processed, so the frame rate is not constant. However, my experience is that as long as you specify a framerate larger than the max achievable frame rate of the source, then the viewer still looks fine. On the flipside, if the video source starts emitting frames at 60 fps, but I specify a framerate of 30, then the delay just sort of builds up in ffplay to the point where after 10 seconds, the video is 20 seconds behind. So the first question would be whether there is a way to get ffplay to use a variable framerate. The behavior I am looking for is "display a frame as soon as it is received over http".

  In light of the aforementioned issue, the approach I have been taking is to simply specify a high framerate, which seems to work. However, there is an issue with this approach in the form of startup time. When I set the framerate to 10, the ffplay window starts in approx 3 seconds, but then quickly starts accruing a lag (so I can’t do this). When I set the framerate to 100, the ffplay window takes 30 seconds (literally 30 seconds) to start, but then will not have any lag.

  I have seen that ffmpeg has a vsync option that on the surface seems like it would allow you to set a variable framerate. However, ffplay doesn’t seem to recognize this. I would also be willing to pipe the output of ffmpeg to a different window (I am running Ubuntu 18.04) if that is what must be done, but I would prefer not to have to recompile ffplay.

• FFMPEG merge audio tracks into one and encode using NVENC

  5 novembre 2019, par L0Lock

  I often shoot films with several audio inputs, resulting in video files with multiple audio tracks supposed to be played all together at the same time.
  I usually go through editing those files and there I do whatever I want with those files, but sometimes I would also like to just send the files right away online without editing, in which case I would enjoy FFMPEG’s fast & simple & quality encoding.

  But here’s the catch : most online video streaming services don’t support multiple audio tracks, so I have to merge them into one so we can hear everything.

  I also want to upscale the video (it’s a little trick for the streaming service to trigger its higher quality encoding).
  And finally, since it’s just an encoding meant to just be shared on a streaming service, I prefer a fast & light encoding over quality, which HEVC NVENC is good for.

  So far I’ve tried to use the amix advanced filter and I try to use the Lanczos filter for upscaling which seems to give a better result in my case.

  The input file is quite simple :

  Stream 0:0 : video track
  Stream 0:1 : main audio recording
  Stream 0:2 : secondary audio recording

  The audio tracks are at the correct volume and duration and position in time, so the only thing I need is really just to turn them into one track

  ffmpeg -i "ow_raw.mp4" -filter_complex "[0:1][0:2]amix=inputs=2[a]" -map "0:0" -map "[a]" -c:v hevc_nvenc -preset fast -level 4.1 -pix_fmt yuv420p -vf scale=2560:1:flags=lanczos "ow_share.mkv" -y

  But it doesn’t work :

  ffmpeg version N-94905-g8efc9fcc56 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 9.1.1 (GCC) 20190807
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  
    libavutil      56. 35.100 / 56. 35.100
  
    libavcodec     58. 56.101 / 58. 56.101
  
    libavformat    58. 32.104 / 58. 32.104
  
    libavdevice    58.  9.100 / 58.  9.100
  
    libavfilter     7. 58.102 /  7. 58.102
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'ow_raw.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: isommp42
  
      creation_time   : 2019-11-02T16:43:32.000000Z
  
      date            : 2019
  
    Duration: 00:15:49.79, start: 0.000000, bitrate: 30194 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, smpte170m/smpte170m/bt470m), 1920x1080 [SAR 1:1 DAR 16:9], 29805 kb/s, 60 fps, 60 tbr, 90k tbn, 120 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-11-02T16:43:32.000000Z
  
        handler_name    : VideoHandle
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 196 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-11-02T16:43:32.000000Z
  
        handler_name    : SoundHandle
  
      Stream #0:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 184 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-11-02T16:43:32.000000Z
  
        handler_name    : SoundHandle
  
  Stream mapping:
  
    Stream #0:1 (aac) -> amix:input0 (graph 0)
  
    Stream #0:2 (aac) -> amix:input1 (graph 0)
  
    Stream #0:0 -> #0:0 (h264 (native) -> hevc (hevc_nvenc))
  
    amix (graph 0) -> Stream #0:1 (libvorbis)
  
  Press [q] to stop, [?] for help
  
  [hevc_nvenc @ 000002287e34a040] InitializeEncoder failed: invalid param (8)
  
  Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  Conversion failed!