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• Multichannel AAC mp4 encoding using libav (avconv) or ffmpeg

  5 octobre 2018, par Josh Kopecek

  I am trying to create a four-channel mp4 file with AAC encoding for ambisonics use. I am trying to encode a 4-channel first-order ambisonic wav file into AAC like so :

  avconv -i four_channel_input.wav -c:a libfaac -ac 4 four_channel_output.mp4

  This gives me the error

  [libfaac @ 0x7f938885a000] Specified channel_layout is not supported
  
  Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  Removing the -ac 4 option gives me a 5 channel file

  Duration: 00:01:21.09, start: 0.021333, bitrate: 218 kb/s
  
  Stream #0:0(und): Audio: aac (LC) [mp4a / 0x6134706D]
  
    48000 Hz, 5.0, fltp, 215 kb/s (default)

  with a blank first channel, which is obviously suboptimal. In order to create compressed ambisonics files, should I be using a separate format like AmbiX (even though I believe this is uncompressed) ?

• avformat/dashdec : drop arbitrary DASH manifest size limit

  3 septembre 2020, par Jan Ekström

  avformat/dashdec : drop arbitrary DASH manifest size limit
  Currently the utilized AVBPrint API is internally limited to unsigned
  
  integers, so if we limit the file size as well as the amount to read
  
  to UINT_MAX - 1, we do not require additional limiting to be performed
  
  on the values.
  This change is based on the fact that initially the 8*1024 value added
  
  in 96d70694aea64616c68db8be306c159c73fb3980 was only for the case where
  
  the file size was not known. It was not a maximum file size limit.
  In 29121188983932f79aef8501652630d322a9974c this was reworked to be
  
  a maximum manifest file size limit, while its commit message appears
  
  to only note that it added support for larger manifest file sizes.
  This should enable various unfortunately large MPEG-DASH manifests,
  
  such as Youtube's multi-megabyte live stream archives to load up
  
  as well as bring back the original intent of the logic.
  

  • [DH] libavformat/dashdec.c

• Transcode to ogg or webm, writing the file as it goes

  22 juillet 2020, par Mark Smith

  I need to transcode files (mp3, flac, m4a and others) to ogg or webm. (This is because I need them to play on Firefox 60.9 which does not support most of these, and flacs are too large. I cannot update the browser.)

  


  ffmpeg can do the transcoding, but when transcoding to ogg or webm, depending on the exact configuration, either 0 bytes or a few kB is written immediately, and then nothing more until the transcoding is complete (even using -flush_packets 1) — hence I cannot start playing the audio.

  


  By comparison, if I transcode to mp3, the file is written progressively and I can start playing immediately.

  


  Is it possible to transcode to ogg or webm in such a way that the file is written as the transcoding happens, and I can start playing it (almost) immediately ?

  


  Configurations I have tried :

  


  ffmpeg -i orig.m4a -c:a libvorbis -flush_packets 1 vorbis.ogg
ffmpeg -i orig.m4a -c:a libopus -flush_packets 1 opus.ogg
ffmpeg -i orig.m4a -c:a libvorbis -flush_packets 1 vorbis.webm
ffmpeg -i orig.m4a -c:a libopus -flush_packets 1 opus.webm


  


  This is running on Debian (Raspian stretch, specifically) and I would like to do it without adding dependencies from outside of the Debian/Raspian archives, if possible. Sticking with ffmpeg would be my ideal choice but will consider others.