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• How can I convince ffserver to save a locally-sourced webcam stream to a file in high resolution AND stream it in lower resolution ?

  2 novembre 2015, par Dominic Jacobssen

  We have a remote Linux machine, accessible over VPN, which has a USB webcam. We want to use this for video conferencing, but we also want to store the stream for archiving.

  Since the streaming bandwidth is limited, it makes sense to capture the stream on the same machine as the webcam and rsync that across after-the-fact, rather than trying to capture the streamed content, which is necessarily going to be poor quality.

  We’re trying to use ffmpeg and ffserver to achieve this, but with little success. Most of the articles on the internet either deal with just streaming a webcam, or rebroadcasting a remote stream. We found we had to recompile ffserver because of a missing "my_addr->sin_family = AF_INET ;" in the version of ffserver.c we had been using, since fixed in git.

  Here’s the ffserver.conf we’re trying to use :

  Port 43688
  
  BindAddress 127.0.0.1
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 1000
  
  CustomLog -
  
  NoDaemon
  
  <feed>
  
  ReadOnlyFile /tmp/feed.ffm
  
  FileMaxSize 20M
  
  ACL allow 127.0.0.1
  
  </feed>
  
  <stream>
  
  Feed feed.ffm
  
  Format mp4
  
  VideoSize qvga
  
  VideoGopSize 12
  
  VideoHighQuality
  
  Video4MotionVector
  
  VideoCodec libx264
  
  VideoBitRate 100
  
  VideoBufferSize 40
  
  VideoFrameRate 5
  
  VideoQMin 3
  
  VideoQMax 31
  
  AudioCodec libfaac
  
  AudioBitRate 32
  
  AudioChannels 2
  
  AudioSampleRate 22050
  
  ACL allow localhost
  
  </stream>

  When we fire this up, we get the error :

  Unable to create feed file '/tmp/feed.ffm' as it is marked readonly

  Fair enough, but this is not what is implied in the docs. Changing the directive to :

  File /tmp/feed.ffm

  allows ffserver to fire up and appear to sit and wait for ffmpeg to connect to it. However, when we fire up ffmpeg with the command :

  ffmpeg -f alsa -i pulse -r 16000 -f video4linux2 -s qvga -i /dev/video0 -r 5 -f mp4 -vcodec libx264 -sameq -acodec libfaac -ab 32k http://127.0.0.1:43688/feed.ffm

  then the webcam lights up and ffserver acknowledges the connection with the messages :

  New connection: POST /feed.ffm
  
  [POST] "/feed.ffm HTTP/1.1" 200 0

  but after a few seconds we get the errors :

  [mp4 @ 0x264b160] muxer does not support non seekable output
  
  Could not write header for output file #0 (incorrect codec parameters ?)

  We’ve tried various other formats (mpeg, mpegts, avi) and codecs (mpeg1video, mpeg2video, mpeg4), all without success.

  Moreover, we were under the impression that ffserver could reencode input format to a lower resolution for streaming, but if the stream resolution doesn’t match the feed resolution, we get an error about the resolutions not matching.

  Has anyone ever managed to get this working correctly ? I’ve read about vlc being able to do something like this, but the vlc command lines are well nigh impenetrable.

  Thanks !

  Dominic

• ffmpeg : AVI to FLV conversion doubles file size

  22 septembre 2012, par weekens

  I convert AVI to FLV with ffmpeg using -sameq parameter (same quality) :

  ffmpeg -i test.avi -sameq -f flv sameq.flv

  The resulting file has the same video and audio quality as the original, but it's more than twice the original file size :

  84M sameq.flv
  
  41M test.avi

  Why does it happen ?

  Transcoder output :

  ffmpeg version N-34750-g070d2d7, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 12 2011 11:23:07 with gcc 4.6.1
  
    configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab
  
    libavutil    51. 24. 1 / 51. 24. 1
  
    libavcodec   53. 33. 0 / 53. 33. 0
  
    libavformat  53. 20. 0 / 53. 20. 0
  
    libavdevice  53.  4. 0 / 53.  4. 0
  
    libavfilter   2. 48. 0 /  2. 48. 0
  
    libswscale    2.  1. 0 /  2.  1. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  Input #0, avi, from 'test.avi':
  
    Duration: 00:06:30.00, start: 0.000000, bitrate: 866 kb/s
  
      Stream #0:0: Video: mpeg4 (Advanced Real Time Simple Profile) (DIVX / 0x58564944), yuv420p, 400x300 [SAR 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc
  
      Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, mono, s16, 64 kb/s
  
  [buffer @ 0xa247ae0] w:400 h:300 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param:
  
  Output #0, flv, to 'sameq.flv':
  
    Metadata:
  
      encoder         : Lavf53.20.0
  
      Stream #0:0: Video: flv1 ([2][0][0][0] / 0x0002), yuv420p, 400x300 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 1k tbn, 25 tbc
  
      Stream #0:1: Audio: mp3 ([2][0][0][0] / 0x0002), 44100 Hz, mono, s16, 128 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mpeg4 -> flv)
  
    Stream #0:1 -> #0:1 (mp3 -> libmp3lame)
  
  Press [q] to stop, [?] for help
  
  frame= 9742 fps=255 q=0.0 Lsize=   85074kB time=00:06:30.00 bitrate=1787.0kbits/s    
  
  video:79163kB audio:5525kB global headers:0kB muxing overhead 0.455568%

• converting of wma to mp3 through ffmpeg is not working

  8 novembre 2011, par Rahul Mehta

  I want to convert my wma file to mp3 by ffmpeg .
  why this error is coming what should i do ? i installed libmp3lame already .

  when i am trying to convert by following command gives error :

  /var/www/yapi$ sudo /usr/local/bin/ffmpeg -i uploads/4e02dc4f197c1.wma uploads/4e02dc4f1b212.mp3
  
  ffmpeg version N-30971-g4b87a08, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Jun 23 2011 10:08:49 with gcc 4.4.5
  
    configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-x11grab
  
    libavutil    51.  9. 1 / 51.  9. 1
  
    libavcodec   53.  7. 0 / 53.  7. 0
  
    libavformat  53.  4. 0 / 53.  4. 0
  
    libavdevice  53.  1. 1 / 53.  1. 1
  
    libavfilter   2. 23. 0 /  2. 23. 0
  
    libswscale    2.  0. 0 /  2.  0. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  Input #0, asf, from 'uploads/4e02dc4f197c1.wma':
  
    Metadata:
  
      title           : "Highway Blues"
  
      artist          : Marc Seales, composer. New Stories. Ernie Watts, saxophone.
  
      genre           : Jazz
  
      album           : Speakin' Out
  
      track           : 1
  
      WM/Year         : 1999
  
      WMFSDKVersion   : 8.00.00.4389
  
      WMFSDKNeeded    : 0.0.0.0000
  
    Duration: 00:01:33.71, start: 0.000000, bitrate: 64 kb/s
  
      Stream #0.0: Audio: wmav2, 44100 Hz, 2 channels, s16, 64 kb/s
  
  File 'uploads/4e02dc4f1b212.mp3' already exists. Overwrite ? [y/N] y
  
  
  
  Output #0, mp3, to 'uploads/4e02dc4f1b212.mp3':
  
      Stream #0.0: Audio: [0][0][0][0] / 0x0000, 44100 Hz, 2 channels, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Encoder (codec id 86017) not found for output stream #0.0

  why it is not working ?