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• ffmpeg video not working in JWPlayer

  20 août 2014, par shailesh gupta

  I have created a video from mixing an audio and a video. By using

   ffmpeg -i video.mp4 -i a.mp3  -vcodec copy -acodec copy -map 0.0 -map 1.0 -shortest output.mp4

  Its working fine in VLC player but no sound in JWPlayer. Its also buffering the whole video before starting the video.
  How to resolve this issue .

  Here is output of ffmpeg command.

   FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  
   built on Jan 29 2012 23:55:02 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  
   configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --    mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-  libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable- libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 -- enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared -- enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  
   libavutil     50.15. 1 / 50.15. 1
  
   libavcodec    52.72. 2 / 52.72. 2
  
   libavformat   52.64. 2 / 52.64. 2
  
   libavdevice   52. 2. 0 / 52. 2. 0
  
   libavfilter    1.19. 0 /  1.19. 0
  
   libswscale     0.11. 0 /  0.11. 0
  
   libpostproc   51. 2. 0 / 51. 2. 0
  
   Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4':
  
   Metadata:
  
     major_brand     : mp42
  
     minor_version   : 0
  
     compatible_brands: isommp42
  
   Duration: 00:01:17.48, start: 0.000000, bitrate: 614 kb/s
  
     Stream #0.0(und): Video: h264, yuv420p, 640x360, 515 kb/s, 29.97 fps, 29.97 tbr, 60k tbn, 59.94 tbc
  
     Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 96 kb/s
  
   [mp3 @ 0x1258e440]max_analyze_duration reached
  
   [mp3 @ 0x1258e440]Estimating duration from bitrate, this may be inaccurate
  
   Input #1, mp3, from 'a.mp3':
  
    Metadata:
  
     TSSE            : Lavf52.64.2
  
    Duration: 00:06:53.52, start: 0.000000, bitrate: 64 kb/s
  
      Stream #1.0: Audio: mp3, 44100 Hz, 2 channels, s16, 64 kb/s
  
   File 'output.mp4' already exists. Overwrite ? [y/N] y
  
   Output #0, mp4, to 'output.mp4':
  
    Metadata:
  
      encoder         : Lavf52.64.2
  
      Stream #0.0(und): Video: libx264, yuv420p, 640x360, q=2-31, 515 kb/s, 60k tbn, 29.97 tbc
  
      Stream #0.1: Audio: libmp3lame, 44100 Hz, 2 channels, 64 kb/s
  
    Stream mapping:
  
      Stream #0.0 -> #0.0
  
      Stream #1.0 -> #0.1
  
    Press [q] to stop encoding
  
    frame= 2322 fps=  0 q=-1.0 Lsize=    5526kB time=77.48 bitrate= 584.3kbits/s    
  
    video:4878kB audio:605kB global headers:0kB muxing overhead 0.777738%

• lavd/v4l2 : do not fail when VIDIOC_ENUMSTD returns ENODATA

  18 août 2014, par Andre Wolokita

  lavd/v4l2 : do not fail when VIDIOC_ENUMSTD returns ENODATA
  As of September 14 2012, v4l_enumstd() will return ENODATA
  
  when a device’s std field is set to 0. That is, the device
  
  does not have a standard format. In order to properly
  
  handle this case, v4l2_set_parameters should catch the
  
  ENODATA code and break instead of failing.
  Below is the v4l2-core commit describing this change.
  >>commit a5338190efc7cfa8c99a6856342a77d21c9a05cf
  
  >>Author : Hans Verkuil <hans.verkuil@cisco.com>
  
  >>Date :   Fri Sep 14 06:45:43 2012 -0300
  
  >>
  
  >>    [media] v4l2-core : tvnorms may be 0 for a given input, handle that case
  
  >>
  
  >>    Currently the core code looks at tvnorms to see whether ENUMSTD
  
  >>    or G_PARM should be enabled. This is not a good check for drivers
  
  >>    that support the STD API on one input and the DV Timings API on another.
  
  >>    In that case tvnorms may be 0.
  
  >>    Instead check whether s_std is present (for ENUMSTD) or whether g_std or
  
  >>    current_norm is present for g_parm.
  
  >>    Also, in the enumstd core function return ENODATA if tvnorms is 0,
  
  >>    because in that case the current input does not support the STD API
  
  >>    and ENUMSTD should return ENODATA for that.
  
  >>
  
  >>    Signed-off-by : Hans Verkuil <hans.verkuil@cisco.com>
  
  >>    Reviewed-by : Sakari Ailus <sakari.ailus@iki.fi>
  
  >>    Signed-off-by : Mauro Carvalho Chehab <mchehab@redhat.com>
  

  • [DH] libavdevice/v4l2.c

• Grep for a number between a certain range - Checking WAV Quality

  11 août 2014, par BT643

  I’m trying to write a regular expression (to be used in conjunction with ffmpeg, which can check that a WAV file is over a certain quality.

  The minimum should be :

  Audio Channels : 2 (Stereo)
  Audio Sample Rate : 44,100 Hz
  Audio Bitrate : 1411 Kbps
  Audio Bit Depth : 16 bit
  

  So I’ve tried the following commands so far :

  • /usr/local/bin/ffmpeg -i "/path/to/file.wav" 2>&1 | egrep 'stereo|2 channels'

  This works fine to get a stero (2 channel) WAV. I’m getting issues with the next part, searching between a range of numbers.

  • /usr/local/bin/ffmpeg -i "/path/to/file.wav" 2>&1 | egrep 'stereo|2 channels' | egrep '[41000-196000] Hz'

  Obviously this just searches each number individually, so it’s finding results if there’s a 4 OR 1 OR 0 OR 0 OR 0 etc...

  The bit rate and bit depth just needs to be OVER 1411 and 16 respectively.

  Thanks

  EDIT -
  Here’s the ffmpeg output for a low quality WAV which should be rejected :

  ffmpeg version git-2012-05-22-27127eb Copyright (c) 2000-2012 the FFmpeg developers
 built on May 22 2012 12:27:21 with gcc 4.6.1
 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab
 libavutil      51. 53.100 / 51. 53.100
 libavcodec     54. 21.101 / 54. 21.101
 libavformat    54.  6.100 / 54.  6.100
 libavdevice    53.  4.100 / 53.  4.100
 libavfilter     2. 75.100 /  2. 75.100
 libswscale      2.  1.100 /  2.  1.100
 libswresample   0. 15.100 /  0. 15.100
 libpostproc    52.  0.100 / 52.  0.100
[wav @ 0x355b140] max_analyze_duration 5000000 reached at 5056000
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from '/path/to/file.wav':
 Duration: 00:02:28.47, bitrate: 512 kb/s
   Stream #0:0: Audio: pcm_u8 ([1][0][0][0] / 0x0001), 32000 Hz, stereo, u8, 512 kb/s