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• avconv : Saving MP4 is Taking Hours

  4 septembre 2015, par PeregrineStudios

  not entirely sure if I should post this on askubuntu or stackoverflow. Posted on askubuntu and not getting any answers. I’m hoping I can find an answer here.

  I’m setting up a website where users can upload videos and share them. I’m using avconv to reduce the video size and save it twice, once as an mp4 and again as a webm.

  Uploading a .MOV from a phone, the video conversion is quick and manageable.

  Uploading a .mp4 from a Samsung Galaxy S3, the video conversion to webm is also quick. But, the conversion to another mp4 takes FOREVER - literally hours. Why ? Is anyone able to shed light on the problem ?

  My avconv output is below.

  avconv -i /var/www/dev_jackpine/ottawa2017/temp_videos/37a949071acaee714dae829ca8df9b29.mp4 -c:v libx264 -vf transpose=1,transpose=1,transpose=1 -s 640x480 /var/www/dev_jackpine/ottawa2017/spinnies/test.mp4
  
  avconv version 0.8.17-4:0.8.17-0ubuntu0.12.04.1, Copyright (c) 2000-2014 the Libav developers
  
  built on Mar 16 2015 13:26:50 with gcc 4.6.3
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/var/www/dev_jackpine/ottawa2017/temp_videos/37a949071acaee714dae829ca8df9b29.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2015-09-04 15:08:21
  
  Duration: 00:00:07.76, start: 0.000000, bitrate: 11756 kb/s
  
  Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720, 11967 kb/s, 29.81 fps, 90k tbr, 90k tbn, 180k tbc
  
  Metadata:
  
    creation_time   : 2015-09-04 15:08:21
  
  Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 119 kb/s
  
  Metadata:
  
    creation_time   : 2015-09-04 15:08:21
  
  File '/var/www/dev_jackpine/ottawa2017/spinnies/test.mp4' already exists. Overwrite ? [y/N] y
  
  [buffer @ 0xc3d580] w:1280 h:720 pixfmt:yuv420p
  
  [scale @ 0xc3dac0] w:1280 h:720 fmt:yuv420p -> w:640 h:480 fmt:yuv420p flags:0x4
  
  [transpose @ 0xc3e280] w:640 h:480 dir:1 -> w:480 h:640 rotation:clockwise vflip:0
  
  [transpose @ 0xc3e7c0] w:480 h:640 dir:1 -> w:640 h:480 rotation:clockwise vflip:0
  
  [transpose @ 0xc3ede0] w:640 h:480 dir:1 -> w:480 h:640 rotation:clockwise vflip:0
  
  [libx264 @ 0xc2b100] MB rate (108000000) > level limit (983040)
  
  [libx264 @ 0xc2b100] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
  
  [libx264 @ 0xc2b100] profile Main, level 5.1
  
  [libx264 @ 0xc2b100] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.25 aq=1:1.00
  
  Output #0, mp4, to '/var/www/dev_jackpine/ottawa2017/spinnies/test.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2015-09-04 15:08:21
  
      encoder         : Lavf53.21.1
  
      Stream #0.0(eng): Video: libx264, yuv420p, 480x640, q=-1--1, 180k tbn, 90k tbc
  
      Metadata:
  
        creation_time   : 2015-09-04 15:08:21
  
      Stream #0.1(eng): Audio: libvo_aacenc, 48000 Hz, stereo, s16, 200 kb/s
  
      Metadata:
  
        creation_time   : 2015-09-04 15:08:21
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 -> libx264)
  
    Stream #0:1 -> #0:1 (aac -> libvo_aacenc)
  
  Press ctrl-c to stop encoding

  Here’s an example frame output :

  frame=124398 fps=142 q=33.0 size=   16885kB time=1.38 bitrate=100110.2kbits/s dup=124356 drop=0

  I’ve had the process running the whole time I’ve been writing this question, and so far it’s done... 22 frames.

  What could be the problem ?

• FFMPEG : Error while opening encoder - when resizing

  10 août 2015, par pufAmuf

  I am trying to resize a video so that the height is 240px.

  Here is the code I used :

  ffmpeg -i input.mp4 -vf scale=-1:240 -acodec mp3 -vcodec libx264 output.mp4

  The error I received was this :

  Error while opening encoder for output stream #0:0 - maybe incorrect
  parameters such as bit_rate, rate, width or height

  Specifically, here is the complete response :

  > [libx264 @ 042e0200] width not divisible by 2 (427x240) Output #0,
  
  > mp4, to 'output_240p.mp4':   Metadata:
  
  >     major_brand     : isom
  
  >     minor_version   : 512
  
  >     compatible_brands: isomiso2avc1mp41
  
  >     encoder         : Lavf53.24.2
  
  >     Stream #0:0(und): Video: h264, none, q=2-31, 128 kb/s, SAR 1280:1281 DAR 0:0 , 25 fps (default)
  
  >     Metadata:
  
  >       creation_time   : 1970-01-01 00:00:00
  
  >       handler_name    : VideoHandler
  
  >       encoder         : Lavc56.57.100 libx264
  
  >     Stream #0:1(und): Audio: mp3, 0 channels, 128 kb/s (default)
  
  >     Metadata:
  
  >       creation_time   : 1970-01-01 00:00:00
  
  >       handler_name    : SoundHandler
  
  >       encoder         : Lavc56.57.100 libmp3lame Stream mapping:   Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))   Stream #0:1 ->
  
  > #0:1 (aac (native) -> mp3 (libmp3lame)) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate,
  
  > rate, width or height

  If I use the below code, it works great, except then I can’t keep on using my aspect ratio :

  ffmpeg -i input.mp4 -s 320x240 -acodec mp3 -vcodec libx264 output.mp4

  Any idea what is going on here ? Thanks !

• "MPEG audio header not found" error when opening with TagLib after converting with ffmpeg

  6 juillet 2015, par Avrohom Yisroel

  I converted a wma file simply by doing something like this...

  ffmpeg -i song.wma -f mp3 song.mp3

  I can then play the mp3 file in Windows Media player, so it looks like the conversion worked.

  However, if I try to open the file in TagLib, I get an error "MPEG audio header not found" on the following line...

  TagLib.File tf = TagLib.File.Create("song.mp3");

  I’ve tried this on a few wma files, so it’s not just that one that’s at fault.

  Anyone have any idea what I did wrong ? I find the docs for ffmpeg pretty overwhelming, and as a complete ignoramus in the field of audio encoding, I haven’t a clue what most of it means. Could be I’m missing something in the conversion, although that wouldn’t explain why WMP can play it but TagLib can’t open it.