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• Stream input to stream output using avserver

  13 janvier 2014, par Gustavo Sgarbi Campos

  I'm trying to use the avserver package to re-stream my ip camera's stream in order to be able to record this stream and access it remotelly without having to stream it twice from the camera, to avoid overloading the wireless network.

  Following this link https://libav.org/avserver.html I came up with the following command :

  avserver -f /etc/avserver.conf & avconv -i http://user:password@192.168.1.121/videostream.asf http://localhost:80/feed1.ffm

  This gives me the following output

  avserver version 0.8.3-6:0.8.3-6ubuntu2, Copyright (c) 2000-2012 the Libav developers
  
  built on Oct  1 2012 15:03:26 with gcc 4.7.2
  
  avconv version 0.8.3-6:0.8.3-6ubuntu2, Copyright (c) 2000-2012 the Libav developers
  
  built on Oct  1 2012 15:03:26 with gcc 4.7.2
  
  Mon Jan 13 20:13:44 2014 AVserver started.
  
  [asf @ 0x153cac0] max_analyze_duration reached
  
  [asf @ 0x153cac0] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, asf, from 'http://admin:gafit@192.168.1.121/videostream.asf':
  
  Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
  Stream #0.0: Video: mjpeg, yuvj422p, 320x240, 100 tbr, 1k tbn, 1k tbc
  
  Stream #0.1: Audio: adpcm_ima_wav, 8000 Hz, 1 channels, s16, 32 kb/s
  
  Incompatible pixel format 'yuvj422p' for codec 'mpeg1video', auto-selecting format 'yuv420p'
  
  [buffer @ 0x153ec80] w:320 h:240 pixfmt:yuvj422p
  
  [avsink @ 0x1544820] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
  
  [scale @ 0x1543460] w:320 h:240 fmt:yuvj422p -> w:320 h:240 fmt:yuv420p flags:0x4
  
  [mp2 @ 0x1544aa0] Requested sampling rate unsupported using closest supported (16000)
  
  Mon Jan 13 20:13:50 2014 Feed '/tmp/feed1.ffm.ffm' stream number does not match registered feed
  
  Output #0, ffm, to 'http://localhost:80/feed1.ffm':
  
  Metadata:
  
  encoder         : Lavf53.21.0
  
  Stream #0.0: Video: mpeg1video, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 1000k tbn, 60 tbc
  
  Stream #0.1: Audio: mp2, 16000 Hz, 1 channels, s16, 128 kb/s
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (mjpeg -> mpeg1video)
  
  Stream #0:1 -> #0:1 (adpcm_ima_wav -> mp2)
  
  Press ctrl-c to stop encoding
  
  Mon Jan 13 20:13:50 2014 127.0.0.1 - - [POST] "/feed1.ffm HTTP/1.1" 200 4096

  When I try to open this stream via VLC, using

  http://192.168.1.15/test.asf

  I get the following additional output

  Mon Jan 13 20:17:39 2014 192.168.1.6 - - [GET] "/test.asf HTTP/1.1" 200 115s dup=12210 drop=0    
  
  Mon Jan 13 20:17:39 2014 Codec for stream 0 does not use global headers but container format requires global headers
  
  Mon Jan 13 20:17:39 2014 Codec for stream 1 does not use global headers but container format requires global headers

  And I get nothing on VLC. I'm not quite sure what is going on or how to fix it, so any enlightning on this would be very welcome.

  Thank you all in advance !

• When capturing input with ffmpeg can't set video bitrate

  11 mai 2014, par Emanuele

  When capturing input from FFmpeg with following command line :
  ffmpeg -f x11grab -r 30 -s 1920x1080 -i :0.0 -vcodec mpeg4 -b:v 10000k -flags +aic+mv4 -f alsa -ac 2 -i pulse -strict experimental test.mp4
  this is the output I get :

  ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
    built on Feb  6 2014 20:56:59 with gcc 4.6.3
  *** THIS PROGRAM IS DEPRECATED ***
  This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
  [x11grab @ 0x1b1c7a0] device : :0.0 -> display : :0.0 x : 0 y : 0 width : 1920 height : 1080
  [x11grab @ 0x1b1c7a0] shared memory extension  found
  [x11grab @ 0x1b1c7a0] Estimating duration from bitrate, this may be inaccurate
  Input #0, x11grab, from ’:0.0’ :
    Duration : N/A, start : 1399805986.592884, bitrate : 1990656 kb/s
      Stream #0.0 : Video : rawvideo, bgra, 1920x1080, 1990656 kb/s, 30 tbr, 1000k tbn, 30 tbc
  [alsa @ 0x1b2c600] capture with some ALSA plugins, especially dsnoop, may hang.
  [alsa @ 0x1b2c600] Estimating duration from bitrate, this may be inaccurate
  Input #1, alsa, from ’pulse’ :
    Duration : N/A, start : 1399805986.561124, bitrate : N/A
      Stream #1.0 : Audio : pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  File ’test.mp4’ already exists. Overwrite ? [y/N] y
  Incompatible pixel format ’bgra’ for codec ’mpeg4’, auto-selecting format ’yuv420p’
  [buffer @ 0x1b4d840] w:1920 h:1080 pixfmt:bgra
  [avsink @ 0x1b40480] auto-inserting filter ’auto-inserted scaler 0’ between the filter ’src’ and the filter ’out’
  [scale @ 0x1b1c6e0] w:1920 h:1080 fmt:bgra -> w:1920 h:1080 fmt:yuv420p flags:0x4
  Output #0, mp4, to ’test.mp4’ :
    Metadata :
      encoder : Lavf53.21.1
      Stream #0.0 : Video : mpeg4, yuv420p, 1920x1080, q=2-31, 200 kb/s, 30 tbn, 30 tbc
      Stream #0.1 : Audio : aac, 48000 Hz, 2 channels, s16, 200 kb/s
  Stream mapping :
    Stream #0.0 -> #0.0
    Stream #1.0 -> #0.1
  Press ctrl-c to stop encoding
  ^Cframe= 1283 fps= 42 q=31.0 Lsize=   10882kB time=42.71 bitrate=2087.3kbits/s dup=473 drop=70    
  video:9875kB audio:980kB global headers:0kB muxing overhead 0.251384%
  Received signal 2 : terminating.
  ema@scv : $ ./capvideo.sh 
  ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
    built on Feb  6 2014 20:56:59 with gcc 4.6.3
  *** THIS PROGRAM IS DEPRECATED ***
  This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
  [x11grab @ 0x24667a0] device : :0.0 -> display : :0.0 x : 0 y : 0 width : 1920 height : 1080
  [x11grab @ 0x24667a0] shared memory extension  found
  [x11grab @ 0x24667a0] Estimating duration from bitrate, this may be inaccurate
  Input #0, x11grab, from ’:0.0’ :
    Duration : N/A, start : 1399806153.224758, bitrate : 1990656 kb/s
      Stream #0.0 : Video : rawvideo, bgra, 1920x1080, 1990656 kb/s, 30 tbr, 1000k tbn, 30 tbc
  [alsa @ 0x2476600] capture with some ALSA plugins, especially dsnoop, may hang.
  [alsa @ 0x2476600] Estimating duration from bitrate, this may be inaccurate
  Input #1, alsa, from ’pulse’ :
    Duration : N/A, start : 1399806153.198090, bitrate : N/A
      Stream #1.0 : Audio : pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  File ’test.mp4’ already exists. Overwrite ? [y/N] y
  Incompatible pixel format ’bgra’ for codec ’mpeg4’, auto-selecting format ’yuv420p’
  [buffer @ 0x2497840] w:1920 h:1080 pixfmt:bgra
  [avsink @ 0x2497c00] auto-inserting filter ’auto-inserted scaler 0’ between the filter ’src’ and the filter ’out’
  [scale @ 0x24a9680] w:1920 h:1080 fmt:bgra -> w:1920 h:1080 fmt:yuv420p flags:0x4
  Output #0, mp4, to ’test.mp4’ :
    Metadata :
      encoder : Lavf53.21.1
      Stream #0.0 : Video : mpeg4, yuv420p, 1920x1080, q=2-31, 200 kb/s, 30 tbn, 30 tbc
      Stream #0.1 : Audio : aac, 48000 Hz, 2 channels, s16, 200 kb/s
  Stream mapping :
    Stream #0.0 -> #0.0
    Stream #1.0 -> #0.1
  Press ctrl-c to stop encoding
  ^Cframe=  229 fps= 36 q=24.8 Lsize=    2316kB time=7.59 bitrate=2497.9kbits/s dup=43 drop=4    
  video:2134kB audio:176kB global headers:0kB muxing overhead 0.253411%
  Received signal 2 : terminating.
  

  As you can see the video is being saved with 200kbs no matter what option I set ; what am I doing wrong ?
  I’m running this on Ubuntu 12.04, ffmpeg is ffmpeg version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers built on Feb  6 2014 20:56:59 with gcc 4.6.3.

  Cheers

• How to join AAC files with AVCONV CONCAT using an input txt file ?

  26 mars 2015, par user27284

  I have a number of AAC files and want to concatenate them into one AAC file. When I do this with AVCONV at the command line and I am specific with the file names, it works. When I try to do this with a text file that holds a list of files, it fails. What led me here is the FFMPEG tutorial page at Concatenating media files. I get a variety of results depending on what I try and I cannot find any info on what I am doing wrong with the syntax.

  My file list (list.txt in the same directory)...

  file 'sr_program_2015_03_23_05_44_01.aac'
  
  file 'sr_program_2015_03_23_07_44_58.aac'

  When I follow the example at the page I mentioned above, I get an error. I included both the use of ffmpeg AND avconv but the results are the same.

  ffmpeg...

  ffmpeg -f concat -i list.txt -c copy output
  
  ffmpeg version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  The ffmpeg program is only provided for script compatibility and will be removed
  
  in a future release. It has been deprecated in the Libav project to allow for
  
  incompatible command line syntax improvements in its replacement called avconv
  
  (see Changelog for details). Please use avconv instead.
  
  Unknown input format: 'concat'

  avconv...

  avconv -f concat -i list.txt -c copy output
  
  avconv version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  Unknown input format: 'concat'

  When I remove the "-f" and use the filenames directly, it works. I have no idea why. The resulting file also plays as expected.

  avconv -i concat:sr_program_2015_03_23_05_44_01.aac\|sr_program_2015_03_23_07_44_58.aac -c copy output.aac
  
  avconv version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  [aac @ 0xcb6cc0] channel element 3.5 is not allocated
  
  [aac @ 0xcb4b20] max_analyze_duration reached
  
  [aac @ 0xcb4b20] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, aac, from 'concat:sr_program_2015_03_23_05_44_01.aac|sr_program_2015_03_23_07_44_58.aac':
  
    Duration: 01:58:34.29, bitrate: 65 kb/s
  
      Stream #0.0: Audio: aac, 44100 Hz, stereo, s16, 65 kb/s
  
  Output #0, adts, to 'output.aac':
  
    Metadata:
  
      encoder         : Lavf53.21.1
  
      Stream #0.0: Audio: aac, 44100 Hz, stereo, 65 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press ctrl-c to stop encoding
  
  size=   57148kB time=7315.03 bitrate=  64.0kbits/s
  
  video:0kB audio:57148kB global headers:0kB muxing overhead 0.000000%

  Noticing that the working approach does not use the ’-f’ option, I tried that again with my first attempt and got a completely different error.

  avconv concat -i list.txt -c copy output.aac
  
  avconv version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  Unable to find a suitable output format for 'concat'

  The reason I want to process these ’broken stream’ aac files from a file list in a file is because I want to create the list from a script/code and then process it as part of a daily automated process. When the streams have broken, there are numerous files with the same date. When all goes well, there is only one file. It happens every other week or so. I want to automate the fix that I usually do manually. I have already accomplished creating the target file list, ironically, from examples on the same page mentioned above when I became stuck with this odd behavior.

  I also want to know what it is I am doing wrong. I see the example I used first everywhere. I have tried this on two different machines running debian but different architectures (arm and x86) and received the same results.

  Also, to make sure I had the latest ffmeg, I compiled it on each system using this page... Compile FFmpeg on Ubuntu, Debian, or Mint

  Thank you for your time.