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• Difference between Deinterlace and Progressive

  21 février 2019, par Sasidharan S

  Is it deinterlace and progressive are same in ffmpeg or different ?

  If it is different how to convert from progressive scan to deinterlace in ffmpeg.

  Code I use :

  ffmpeg -ss 00:00:00.000 -t 00:01:00.000 -i 1.mp4 -ss 00:01:00.001 -t 00:02:00.000 -i 2.mp4 -filter_complex "[0:v:0] [1:v:0] concat=n=2:v=1[v] " -map "[v]" -c:v libx264 -preset fast -profile:v high -level 4.1 -coder 1 -yadif -pix_fmt yuv420p -g 50 -bf 3 -x264-params "keyint_min=50:sc_threshold=0:nal-hrd=cbr:subq=6:motion-est=1:trellis=2:8x8dct=1:b-pyramid=0" -b:v:0 500k -minrate 500k -maxrate 500k -bufsize 100k -s:v:0 512x384 -f mp4 -threads 4 -filter_complex "[0:a] [1:a] concat=n=2:v=0:a=1[a]" -map "[a]" -ar 48000 -ac 2 -b:a 128K -write_tmcd off SD_4X3_500k_1.mp4

• MP3 files created using FFmpeg are not starting playback in browser immediately. Is there any major difference between FFmpeg and AVCONV ?

  23 janvier 2019, par AR5

  I am working on a website that streams music. We recently changed server from Debian (with avconv) to a Centos7 (with FFmpeg) server.
  The mp3 files created on Debian server start playback on browser (I have tested Chrome and Firefox) start almost at the same time they start loading into the browser (I used Network tab on Developer Tools to monitor this)

  Now after the switch to Centos/FFmpeg server, the files being created on this new server are displaying a strange behavior. They only start playback after about 1MB is loaded into the browser.

  I have used identical settings for converting original file into MP3 in both AVCONV and FFmpeg but the files created using FFmpeg are showing this issue. Is there something that might be causing such an issue ? Are there differences in terms of audio conversion between AVCONV and FFmpeg ?

  I have already tried

  I first found that the files created on old server (Debian/Avconv) were VBR (variable bitrate) and the ones created on new server were CBR (constant bitrate), so I tried switching to VBR but the issue still persisted.

  I checked the mp3 files using MediaInfo app and there seems to be no difference between the files.

  I also checked if both files were being served as 206 Partial Content and they both are indeed.

  I am trying to create mp3 files using FFmpeg that work exactly like the ones created before using avconv

  I am trying to make the streaming site work on the new server but the mp3 files created using FFmpeg are not playing back correctly as compared to the ones created on the old server. I am trying to figure out what I might be doing wrong ? or if there is a difference between avconv and FFmpeg that is causing this issue.

  I am really stuck on this issue, any help will be really appreciated.

  ---

  Edit

  I don’t have access to old server anymore so I couldn’t retrieve the log output of avconv. The command that I was using was as follows :

  avconv -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"

  Here is the command and log output from new server :

  ffmpeg -y -i "/test/Track 01.mp3" -ac 2 -ar 44100 -acodec libmp3lame -b:a 128k "/test/Track 01 (converted).mp3"
  
  ffmpeg version 2.8.15 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0xd60be0] Skipping 0 bytes of junk at 240044.
  
  Input #0, mp3, from '/test/Track 01.mp3':
  
    Metadata:
  
      album           : Future Hndrxx Presents: The WIZRD
  
      artist          : Future
  
      genre           : Hip-Hop
  
      title           : Never Stop
  
      track           : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      date            : 2019
  
      encoder         : Lavf56.40.101
  
    Duration: 00:04:51.40, start: 0.025056, bitrate: 121 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 114 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60
  
      Stream #0:1: Video: png, rgb24(pc), 333x333 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
  [mp3 @ 0xd66ec0] Frame rate very high for a muxer not efficiently supporting it.
  
  Please consider specifying a lower framerate, a different muxer or -vsync 2
  
  Output #0, mp3, to '/test/Track 01 (converted).mp3':
  
    Metadata:
  
      TALB            : Future Hndrxx Presents: The WIZRD
  
      TPE1            : Future
  
      TCON            : Hip-Hop
  
      TIT2            : Never Stop
  
      TRCK            : 1
  
      lyrics-eng      : rgf.is
  
      WEB SITE        : rgf.is
  
      TAGGINGTIME     : rgf.is
  
      WEB             : rgf.is
  
      TDRC            : 2019
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Video: png, rgb24, 333x333 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
  
      Metadata:
  
        comment         : Cover (front)
  
        encoder         : Lavc56.60.100 png
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (png (native) -> png (native))
  
    Stream #0:0 -> #0:1 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0xd9b0c0] Trying to remove 1152 samples, but the queue is emptys/s
  
  frame=    1 fps=0.1 q=-0.0 Lsize=    4788kB time=00:04:51.39 bitrate= 134.6kbits/s
  
  video:234kB audio:4553kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.014809%

  Samples of MP3 files

  I have uploaded samples of mp3 files created using both avconv and FFmpeg. Please find these here : https://drive.google.com/drive/folders/1gRTmMM2iSK0VWQ4Zaf_iBNQe5laFJl08?usp=sharing

• FFMpeg sws_scale Static and Shared Huge Performance Difference

  6 novembre 2018, par Ali

  I used swscale in my code as a shared library then managed to compile FFMpeg (4.1) to static libraries with Visual Studio with this command just to get swscale :

  ./configure --toolchain=msvc --arch=x86_32 --disable-everything --disable-programs 

  I have nasm and yasm installed. this my config output :

  install prefix            /usr/local
  
  source path               .
  
  C compiler                cl
  
  C library                 msvcrt
  
  ARCH                      x86 (generic)
  
  big-endian                no
  
  runtime cpu detection     yes
  
  standalone assembly       yes
  
  x86 assembler             nasm
  
  MMX enabled               yes
  
  MMXEXT enabled            yes
  
  3DNow! enabled            yes
  
  3DNow! extended enabled   yes
  
  SSE enabled               yes
  
  SSSE3 enabled             yes
  
  AESNI enabled             yes
  
  AVX enabled               yes
  
  AVX2 enabled              yes
  
  AVX-512 enabled           yes
  
  XOP enabled               yes
  
  FMA3 enabled              yes
  
  FMA4 enabled              yes
  
  i686 features enabled     yes
  
  CMOV is fast              no
  
  EBX available             no
  
  EBP available             no
  
  debug symbols             yes
  
  strip symbols             no
  
  optimize for size         no
  
  optimizations             yes
  
  static                    yes
  
  shared                    no
  
  postprocessing support    no
  
  network support           yes
  
  threading support         w32threads
  
  safe bitstream reader     yes
  
  texi2html enabled         no
  
  perl enabled              no
  
  pod2man enabled           no
  
  makeinfo enabled          no
  
  makeinfo supports HTML    no
  
  
  
  External libraries:
  
  schannel
  
  
  
  External libraries providing hardware acceleration:
  
  d3d11va                    dxva2
  
  
  
  Libraries:
  
  avcodec                    avdevice                   avfilter                   avformat                   avutil                     swresample                 swscale
  
  
  
  Programs:
  
  Enabled decoders:
  
  Enabled encoders:
  
  Enabled hwaccels:
  
  Enabled parsers:
  
  Enabled demuxers:
  
  Enabled muxers:
  
  Enabled protocols:
  
  Enabled filters:
  
  Enabled bsfs:
  
  null
  
  Enabled indevs:
  
  Enabled outdevs:

  This compiled successfully and I replaced the lib files with .a file in QT :

  INCLUDEPATH += $$PWD/ffmpeg/inc/
  
  LIBS += $$files($$PWD/ffmpeg/lib/*.a, true)

  I didn’t change anything else. EXE works correctly without dependency but problem is static swscale is so much slower than the shared one. For 1080p share .DLL takes 2ms to shrink and convert yuv to rgb and static .A takes 6ms to

  I also tried removing --disable-everything --disable-programs but still the same. I want to know if it’s because of the cl compiler or I missed a library or a setting ?

  BTW this my system : Win10/i7 4820K/16GB/GTX970

  EDIT :

  I got this in app output :
  No accelerated colorspace conversion found from yuv420p to bgra.
  Although x86 folder in swscale is compiled, it seems it’s not linked in the output.