Recherche avancée

Sur d’autres sites (8831)

• avcodec/vc2enc : Don't use bitcount when byte-aligned

  6 octobre 2022, par Andreas Rheinhardt

  avcodec/vc2enc : Don't use bitcount when byte-aligned
  (There is a small issue that is now being treated differently :
  
  The earlier code would record a position in a buffer that
  
  is being written to via put_bits(), then write data,
  
  then overwrite the byte at the position recorded earlier
  
  and only then flush the PutBitContext. In case there was
  
  no writeout in the meantime, said flush would overwrite
  
  what one has just written. This never happened in my tests,
  
  but maybe it can happen. In this case this commit fixes
  
  this issue by flushing before overwriting the old data.)
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavcodec/vc2enc.c

• Compile FFmpeg with libfdk_aac

  18 février 2017, par Toydor

  I been reading on how to convert mp3 to m4a, and found that I must compile FFmpeg if I’ll use the AAC encoder, libfdk_aac.

  But reading FFmpeg guide on how to compile FFmpeg with libfdk_aac makes no sense for a beginner like me.

  To use libfdk_aac the encoding guide says :

  Requires ffmpeg to be configured with —enable-libfdk_aac
  —enable-nonfree.

  Where do I put those flags ?

  Do I put it here somewhere ? :

  cd ~/ffmpeg_sources
  
  git clone --depth 1 git://github.com/mstorsjo/fdk-aac.git
  
  cd fdk-aac
  
  autoreconf -fiv
  
  ./configure --prefix="$HOME/ffmpeg_build" --disable-shared
  
  make
  
  make install
  
  make distclean

  Or maybe here somewhere ?

  cd ~/ffmpeg_sources
  
  git clone --depth 1 git://source.ffmpeg.org/ffmpeg
  
  cd ffmpeg
  
  PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig"
  
  export PKG_CONFIG_PATH
  
  ./configure --prefix="$HOME/ffmpeg_build" \
  
    --extra-cflags="-I$HOME/ffmpeg_build/include" --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  
    --bindir="$HOME/bin" --extra-libs="-ldl" --enable-gpl --enable-libass --enable-libfdk-aac \
  
    --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx \
  
    --enable-libx264 --enable-nonfree --enable-x11grab
  
  make
  
  make install
  
  make distclean
  
  hash -r

  If I’m reading the compile guide right I guess that these two chunks of code is what I need to compile FFmpeg.

  I’m using Ubuntu server 12.4

  UPDATE

  After upgrading my system to Ubuntu 16.04 I had to install ffmpeg again.
  I still needed libfdk-aac. Fortunately there’s a good step-by-step guide at http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu on how to compile ffmpeg.

  I thought I would share how to compile if just interested in compiling ffmpeg with libfdk-aac and libmp3lame.

  If you haven’t already a bin in home directory :

  mkdir ~/bin 

  Install dependencies. Didn’t need the non-server packages :

  sudo apt-get update
  
  sudo apt-get -y install autoconf automake build-essential libass-dev libfreetype6-dev libtheora-dev libtool libvorbis-dev pkg-config texinfo zlib1g-dev 

  Then install the encoders. Had to install yasm as well, otherwise I got errors when compiling.

  sudo apt-get install libfdk-aac-dev
  
  sudo apt-get install libmp3lame-dev
  
  sudo apt-get install yasm

  Then compile ffmpeg with needed flags

  cd ~/ffmpeg_sources
  
  wget http://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2
  
  tar xjvf ffmpeg-snapshot.tar.bz2
  
  cd ffmpeg
  
  PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
  
  --prefix="$HOME/ffmpeg_build" \
  
  --pkg-config-flags="--static" \
  
  --extra-cflags="-I$HOME/ffmpeg_build/include" \
  
  --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  
  --bindir="$HOME/bin" \
  
  --enable-libass \
  
  --enable-libfdk-aac \
  
  --enable-libfreetype \
  
  --enable-libtheora \
  
  --enable-libvorbis \
  
  --enable-libmp3lame \
  
  --enable-nonfree \
  
  --enable-gpl
  
  PATH="$HOME/bin:$PATH" make
  
  make install
  
  make distclean
  
  hash -r

• Convert m4a to wav, but respect edit list ffmpeg [closed]

  4 février 2024, par user3413723

  I'm using ffmpeg to convert a m4a file to wav. The problem it the m4a file has an edts list that makes the audio start after a bit. I need this to be included so the wav is silent for that time. The audio has to be synced correctly. Any ideas ? I tried these :

  


  ffmpeg -i short.m4a  -acodec pcm_s16le -ac 1 -ar 16000 -use_editlist 1 out.wav

ffmpeg -i short.m4a  -acodec pcm_s16le -ac 1 -ar 16000 -advanced_editlist 0 out.wav


  


  no luck with either. Any ideas ? Here is the file

  


  https://drive.google.com/file/d/10x3QFjT_67ByfgXtiNFE1RlLAcwG68e5/view?usp=sharing

  


  Here is the debugger where you can see it adds time at the beginning. Just upload the m4a file and you can see the info about the audio track. You can also see it if you do ffprobe file.m4a like this :  Duration: 00:00:14.84, start: 0.167007, bitrate: 228 kb/s

  


  mp4 inspector :
https://gpac.github.io/mp4box.js/test/filereader.html