title: ffmpeg coversion for rtmp stream, fname empty
body: <p>I want to use HLS_Variant feature for the NGINX rtmp Module.
  But if I follow the examples in the documentation I can't get it to work.</p>
  <p>I have the following test:</p>
  <pre><code>application Test {
                  live on;
                  record off;
  
                  on_publish http://127.0.0.1/php/rtmp_auth.php;
                  on_publish_done http://127.0.0.1/php/on_publish_done.php;
  
                  exec_push /usr/local/bin/ffmpeg -loglevel debug -i rtmp://localhost:1935/Test/$name
                 -c:v libx264 -acodec aac -preset veryfast -b:v 256k -tune zerolatency -vf &quot;scale=480:trunc(ow/a/2)*2&quot; -f flv  rtmp://localhost:1935/Z006/$name_low
                 -c:v libx264 -acodec aac -preset veryfast -b:v 768k -tune zerolatency -vf &quot;scale=720:trunc(ow/a/2)*2&quot; -f flv  rtmp://localhost:1935/Z006/$name_mid
                 -c:v libx264 -acodec aac -preset veryfast -b:v 1024k -tune zerolatency -vf &quot;scale=960:trunc(ow/a/2)*2&quot; -f flv  rtmp://localhost:1935/Z006/$name_high
                 -c:v libx264 -acodec aac -preset veryfast -b:v 1920k -tune zerolatency -vf &quot;scale=1280:trunc(ow/a/2)*2&quot; -f flv  rtmp://localhost:1935/Z006/$name_higher
                 -c copy -f flv rtmp://localhost:1935/Z006/$name_src 2&gt;&gt;/tmp/Log.log; }
  
          application Z006 {
                  live on;
                  record off;
  
                  hls on;
                  hls_path /usr/local/www/stream/tmp/hls0;
                  hls_nested on;
  
                  hls_variant _low BANDWIDTH=288000;      # _low          - Low bitrate, sub-SD resolution
                  hls_variant _mid BANDWIDTH=448000;      # _mid          - Medium bitrate, SD resolution
                  hls_variant _high BANDWIDTH=1152000;    # _high         - Higher-than-SD resolution
                  hls_variant _higher BANDWIDTH=2048000;  # _higher       - High bitrate, HD 720p resolution
                  hls_variant _src BANDWIDTH=4096000;     # _src          - Source bitrate, source resolution 
  }
  </code></pre>
  <p>As you can see there is nothing special about that.
  The generated log looks like this:</p>
  <pre><code>ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers
    built with FreeBSD clang version 10.0.1 (git@github.com:llvm/llvm-project.git llvmorg-10.0.1-0-gef32c611aa2)
    configuration: --prefix=/usr/local --mandir=/usr/local/man --datadir=/usr/local/share/ffmpeg --pkgconfigdir=/usr/local/libdata/pkgconfig --disable-static --enable-shared --enable-pic --enable-gpl --enable-avresample --cc=cc --cxx=c++ -&gt;
    libavutil      56. 70.100 / 56. 70.100
    libavcodec     58.134.100 / 58.134.100
    libavformat    58. 76.100 / 58. 76.100
    libavdevice    58. 13.100 / 58. 13.100
    libavfilter     7.110.100 /  7.110.100
    libavresample   4.  0.  0 /  4.  0.  0
    libswscale      5.  9.100 /  5.  9.100
    libswresample   3.  9.100 /  3.  9.100
    libpostproc    55.  9.100 / 55.  9.100
  Splitting the commandline.
  Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
  Reading option '-i' ... matched as input url with argument 'rtmp://localhost:1935/Test//'.
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'libx264'.
  Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
  Reading option '-preset' ... matched as AVOption 'preset' with argument 'veryfast'.
  Reading option '-b:v' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '256k'.
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  Reading option '-vf' ... matched as option 'vf' (set video filters) with argument 'scale=480:trunc(ow/a/2)*2'.
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  Reading option 'rtmp://localhost:1935/Z006//_low' ... matched as output url.
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'libx264'.
  Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
  Reading option '-preset' ... matched as AVOption 'preset' with argument 'veryfast'.
  Reading option '-b:v' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '768k'.
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  Reading option '-vf' ... matched as option 'vf' (set video filters) with argument 'scale=720:trunc(ow/a/2)*2'.
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  Reading option 'rtmp://localhost:1935/Z006//_mid' ... matched as output url.
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'libx264'.
  Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
  Reading option '-preset' ... matched as AVOption 'preset' with argument 'veryfast'.
  Reading option '-b:v' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '1024k'.
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  Reading option '-vf' ... matched as option 'vf' (set video filters) with argument 'scale=960:trunc(ow/a/2)*2'.
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  Reading option 'rtmp://localhost:1935/Z006//_high' ... matched as output url.
  Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'libx264'.
  Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
  Reading option '-preset' ... matched as AVOption 'preset' with argument 'veryfast'.
  Reading option '-b:v' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '1920k'.
  Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
  Reading option '-vf' ... matched as option 'vf' (set video filters) with argument 'scale=1280:trunc(ow/a/2)*2'.
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  Reading option 'rtmp://localhost:1935/Z006//_higher' ... matched as output url.
  Reading option '-c' ... matched as option 'c' (codec name) with argument 'copy'.
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  Reading option 'rtmp://localhost:1935///_src' ... matched as output url.
  Finished splitting the commandline.
  Parsing a group of options: global .
  Applying option loglevel (set logging level) with argument debug.
  Successfully parsed a group of options.
  Parsing a group of options: input url rtmp://localhost:1935/Test//.
  Successfully parsed a group of options.
  Opening an input file: rtmp://localhost:1935/Test//.
  [NULL @ 0x80609a000] Opening 'rtmp://localhost:1935/Test//' for reading
  [rtmp @ 0x806089300] No default whitelist set
  [tcp @ 0x806089380] No default whitelist set
  [tcp @ 0x806089380] Original list of addresses:
  [tcp @ 0x806089380] Address 127.0.0.1 port 1935
  [tcp @ 0x806089380] Address ::1 port 1935
  [tcp @ 0x806089380] Interleaved list of addresses:
  [tcp @ 0x806089380] Address 127.0.0.1 port 1935
  [tcp @ 0x806089380] Address ::1 port 1935
  [tcp @ 0x806089380] Starting connection attempt to 127.0.0.1 port 1935
  [tcp @ 0x806089380] Successfully connected to 127.0.0.1 port 1935
  [rtmp @ 0x806089300] Handshaking...
  [rtmp @ 0x806089300] Type answer 3
  [rtmp @ 0x806089300] Server version 13.14.10.13
  [rtmp @ 0x806089300] Proto = rtmp, path = /Test//, app = Test/, fname =
  [rtmp @ 0x806089300] Window acknowledgement size = 5000000
  [rtmp @ 0x806089300] Max sent, unacked = 5000000
  [rtmp @ 0x806089300] New incoming chunk size = 4096
  [rtmp @ 0x806089300] Creating stream...
  [rtmp @ 0x806089300] Sending play command for ''
  </code></pre>
  <p>I think the error is clearly the Empty 'fname'.
  But I don't know what to do about that?</p>
  <p>EDIT:</p>
  <p>Even if I change $name to my Streamname which I use in OBS the conversion isn't started.</p>