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• correctly escaping text in drawtext filter

  6 novembre 2013, par mente

  According to documentation %{}:, should be escaped with slash. But when I escape % the text is not added. There's error message saying Stray % near ')'. But what does it mean and how I can fix it? Command and output:

  $ /usr/bin/ffmpeg -y -i /home/www/255871.mov -af 'aresample=async=1:min_hard_comp=0.100000' -vf 'scale=480:trunc(ow/a/2)*2,transpose=1 [in];movie=watermarks/text-box440.mov,scale=320:55 [bg];[in][bg] overlay=x=20:y=main_h-80,drawtext=fontsize=20:fix_bounds=1:fontfile=Helvetica\\:Style=bold:fontcolor=0xe6b300:text=small kitten \%\):x=30:y=h-80+ascent/2:draw=gte(t\,0)\;lt(t\,5) [out]' -pix_fmt yuv420p -r 24.39 -map_metadata -1 -movflags +faststart -f mp4 -vcodec libx264 -b:v 800k -maxrate 1100k -bufsize 2M -bt 256k -profile:vbaseline -acodec libfdk_aac -ab 64k -ar 44100 -ac 1 /tmp/kitten.mp4
  ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
    built on Mar 15 2013 18:40:14 with gcc 4.4.6 (GCC) 20120305 (Red Hat 4.4.6-4)
    configuration: --enable-gpl --enable-libmp3lame --enable-libvo-aacenc --enable-libx264 --enable-version3 --enable-pthreads --enable-libfaac --enable-nonfree --enable-libfdk_a
  ac --enable-fontconfig --enable-libfreetype
    libavutil      52. 18.100 / 52. 18.100
    libavcodec     54. 92.100 / 54. 92.100
    libavformat    54. 63.104 / 54. 63.104
    libavdevice    54.  3.103 / 54.  3.103
    libavfilter     3. 42.103 /  3. 42.103
    libswscale      2.  2.100 /  2.  2.100
    libswresample   0. 17.102 /  0. 17.102
    libpostproc    52.  2.100 / 52.  2.100
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/www/255871.mov':
    Metadata:
      major_brand     : qt
      minor_version   : 0
      compatible_brands: qt
      creation_time   : 2012-07-17 07:42:35
      model           : iPhone 3GS
      model-rus       : iPhone 3GS
      encoder         : 4.2.1
      encoder-rus     : 4.2.1
      date            : 2012-07-17T10:42:35+0300
      date-rus        : 2012-07-17T10:42:35+0300
      make            : Apple
      make-rus        : Apple
    Duration: 00:00:09.38, start: 0.000000, bitrate: 790 kb/s
      Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 480x360, 720 kb/s, 29.97 fps, 29.97 tbr, 600 tbn, 1200 tbc
      Metadata:
        rotate          : 90
        creation_time   : 2012-07-17 07:42:35
        handler_name    : Core Media Data Handler
      Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 63 kb/s
      Metadata:
        creation_time   : 2012-07-17 07:42:35
        handler_name    : Core Media Data Handler
  [Parsed_drawtext_5 @ 0x2ae9b00] Using "/usr/share/fonts/default/Type1/n019003l.pfb"
  [libx264 @ 0x2ae7d40] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  [libx264 @ 0x2ae7d40] profile Constrained Baseline, level 2.1
  [libx264 @ 0x2ae7d40] 264 - core 130 r2274 c832fe9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=24 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=800 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1100 vbv_bufsize=2000 nal_hrd=none ip_ratio=1.40 aq=1:1.00
  Output #0, mp4, to '/tmp/kitten.mp4':
    Metadata:
      encoder         : Lavf54.63.104
      Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 360x480, q=-1--1, 800 kb/s, 19512 tbn, 24.39 tbc
      Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, mono, s16, 64 kb/s
  Stream mapping:
    Stream #0:0 -> #0:0 (h264 -> libx264)
    Stream #0:1 -> #0:1 (aac -> libfdk_aac)
  Press [q] to stop, [?] for help
  [Parsed_drawtext_5 @ 0x2ae9b00] Stray % near ')'
      Last message repeated 201 times
  Stray % near ')'0.0 q=27.0 size=     480kB time=00:00:07.96 bitrate= 493.4kbits/s dup=0 drop=35
  [Parsed_drawtext_5 @ 0x2ae9b00] Stray % near ')'
      Last message repeated 77 times
  Starting second pass: moving header on top of the file
  frame=  231 fps=0.0 q=-1.0 Lsize=    1002kB time=00:00:09.47 bitrate= 866.6kbits/s dup=0 drop=50
  video:921kB audio:74kB subtitle:0 global headers:0kB muxing overhead 0.680957%
  [libx264 @ 0x2ae7d40] frame I:1     Avg QP:24.43  size: 13264
  [libx264 @ 0x2ae7d40] frame P:230   Avg QP:23.72  size:  4041
  [libx264 @ 0x2ae7d40] mb I  I16..4: 24.8%  0.0% 75.2%
  [libx264 @ 0x2ae7d40] mb P  I16..4:  2.3%  0.0%  4.8%  P16..4: 34.4% 28.8%  9.4%  0.0%  0.0%    skip:20.3%
  [libx264 @ 0x2ae7d40] coded y,uvDC,uvAC intra: 62.5% 54.2% 4.4% inter: 19.6% 20.3% 0.1%
  [libx264 @ 0x2ae7d40] i16 v,h,dc,p: 36% 29% 19% 16%
  [libx264 @ 0x2ae7d40] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 21% 20%  6%  8%  7%  7%  6%  6%
  [libx264 @ 0x2ae7d40] i8c dc,h,v,p: 65% 15% 17%  3%
  [libx264 @ 0x2ae7d40] ref P L0: 93.4%  4.3%  2.3%
  [libx264 @ 0x2ae7d40] kb/s:796.32
  

• ffmpeg libavcodec.so missing while compile with cygwin

  6 novembre 2013, par nick

  I am building ffmpeg for android by following this tutorial

  now i got android folder inside the ffmpeg2.0.1 folder but there is no libavcodec-55.so file. instead of that i have lib/libavcodec.a

  how can i get libavcodec.so file? build_android.sh

  #!/bin/bash
  NDK=$HOME/Desktop/adt/android-ndk-r9
  SYSROOT=$NDK/platforms/android-9/arch-arm/
  TOOLCHAIN=$NDK/toolchains/arm-linux-androideabi-4.8/prebuilt/windows-x86_64
  function build_one
  {
  ./configure \
  --prefix=$PREFIX \
  --enable-shared \
  --disable-static \
  --disable-doc \
  --disable-ffmpeg \
  --disable-ffplay \
  --disable-ffprobe \
  --disable-ffserver \
  --disable-avdevice \
  --disable-doc \
  --disable-symver \
  --cross-prefix=$TOOLCHAIN/bin/arm-linux-androideabi- \
  --target-os=linux \
  --arch=arm \
  --enable-cross-compile \
  --sysroot=$SYSROOT \
  --extra-cflags="-Os -fpic $ADDI_CFLAGS" \
  --extra-ldflags="$ADDI_LDFLAGS" \
  $ADDITIONAL_CONFIGURE_FLAG
  make clean
  make
  make install
  }
  CPU=arm
  PREFIX=$(pwd)/android/$CPU 
  ADDI_CFLAGS="-marm"
  build_one
  

• ffmpeg audio conversion distorted - half rate

  6 novembre 2013, par user1688971

  I'm trying to convert an asf audio to mp3 using ffmpeg. But I have one specific audio that gets distorted in the middle and starts like if the person was talking in slow motion (at half rate).

  The command I'm using is:

  ffmpeg - i input.asf -ac 2 output.mp3

  I've tried a lot of options, but about the middle of the audio is when it fails. The raw file sounds good, so it's not the recording. It is af in the middle of the transmission the frame rate went down for some reason.

  Thanks all!

  [EDIT]

  I'm adding the console response after running the command as a suggestion from LordNeckbeard:

  [root@mynasserver home]# ffmpeg -i recording-8532-1.asf -ac 2 -ab 64k -ar 44100 recording-8532-ac2-ar44100.mp3
  FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --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 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --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
  [flv @ 0x86a4850]max_analyze_duration reached
  [flv @ 0x86a4850]Estimating duration from bitrate, this may be inaccurate
  Input #0, flv, from 'recording-8532-1.asf':
  Metadata:
  source          : STW MediaProxy v3.3.7.19894
  Duration: 04:00:08.49, start: 0.000000, bitrate: N/A
  Stream #0.0: Audio: aac, 44100 Hz, 2 channels (FC), s16
  Output #0, mp3, to 'recording-8532-ac2-ar44100.mp3':
  Metadata:
  TSSE            : Lavf52.64.2
  Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  Stream mapping:
  Stream #0.0 -> #0.0
  Press [q] to stop encoding
  size=  150906kB time=19315.93 bitrate=  64.0kbits/s    
  video:0kB audio:150906kB global headers:0kB muxing overhead 0.000021%
  

  So from the data above, you can see the input file is about 4hrs. The output ends up being around 5 hrs 20 mins.

• bash script fail when a command wait ffmpeg to finish

  5 novembre 2013, par d4r10

  In ubuntu 13.04 have a node.js server that launch for certain events, a script like this:

  #! /bin/bash
  ...
  ...
  
  avconv -i "$dir$video" -ss "$ss.000" -t $tt -threads 0 "$folder/$dif.webm"
  
  y_uid=$(/app/youtube-uploader "$folder/$dif.webm" "$label" "$desc" | grep 'watch' | sed 's/v=/ /g' | awk '{print$2}')
  
  echo "$y_uid" >> ~/log/yt.log
  ...
  ...
  

  if avconv (ffmpeg) process a small file (less then 30 minutes of encoding), the script works fine, but if avconv runs for hours, the next command "...youtube-uploader..." (that is a python based application) doesn't work (but if i try to launch it manually, it will go fine).

  I have also tried to launch the next command inside a loop that check if the variable "$y_uid" is empty, so it will try to repeat youtube-uploader until it find a right value, but it will continue to stay in background and will try forever.. (and it is strange, because if i try to launch the command manually, it works fine..).

  Summarizing, seems like that if avconv run for hours, the avconv process will generate a file that is not recognised or locked or a ghost file for the same script (but is recognised if the command is launched outside the script), if it is a small encoding, it will work fine.

  Unfortunatelly It is difficult to debug or test, because i have every time to wait for hours, please help!

  UPDATE

  I think the problem is related at the buffer size of node.js OR some other buffer that store the standard output of ffmpeg. Just correct

  avconv -i "$dir$video" -ss "$ss.000" -t $tt -threads 0 "$folder/$dif.webm"
  

  in

  avconv -i "$dir$video" -ss "$ss.000" -t $tt -threads 0 "$folder/$dif.webm" &> /dev/null
  

  so all the ffmpeg output not full the buffer memory and everythings go well

• ffmpeg "select" filter not working with MPEG-4 and "-vcodec copy"

  5 novembre 2013, par Simone Palazzo

  I have an MPEG-4 video, which mediainfo describes as:

  Format                                   : MPEG-4
  Format profile                           : Base Media
  Codec ID                                 : isom
  Overall bit rate                         : 6 772 Kbps
  Writing application                      : Lavf54.6.100
  
  Video
  ID                                       : 1
  Format                                   : AVC
  Format/Info                              : Advanced Video Codec
  Format profile                           : High@L4.1
  Format settings, CABAC                   : Yes
  Format settings, ReFrames                : 4 frames
  Codec ID                                 : avc1
  Codec ID/Info                            : Advanced Video Coding
  Bit rate                                 : 25.0 Mbps
  Frame rate mode                          : Constant
  Frame rate                               : 29.000 fps
  Color space                              : YUV
  Chroma subsampling                       : 4:2:0
  Bit depth                                : 8 bits
  Scan type                                : Progressive
  Bits/(Pixel*Frame)                       : 0.554
  Stream size                              : 500 MiB (98%)
  Writing library                          : x264 core 125
  Encoding settings                        : cabac=1 / ref=3 / deblock=1:0:0 /     nalyse=0x3:0x113 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=12 / lookahead_threads=2 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=2 / keyint=250 / keyint_min=25 / scenecut=40 / intra_refresh=0 / rc_lookahead=40 / rc=abr / mbtree=1 / bitrate=25000 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00
  

  (I removed some useless information)

  What I'd like to do is remove the first, say, 200 frames from the video.

  However, if I try with ffmpeg (version 0.11.1, compiled from source)

  ffmpeg -i video.mp4 -vf select='gte(n\,200)' -vcodec copy -strict -2 out.mp4
  

  it doesn't work, the frames are exactly the same. If I remove the -vcodec copy part, what happens is that the first 190 frames are all the same (equal to the 190th one); so, they're not removed, but replaced with the 190th one.

  If I try the select filter to extract single images from the video, with

  ffmpeg -i video.mp4 -vf "select=gte(n\,200)" %d.jpg
  

  it works alright.

  Does anybody have an idea why this happens?

  Thank you!