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• lavf : Add an MPEG-DASH ISOFF segmenting muxer

  6 octobre 2014, par Martin Storsjö

  lavf : Add an MPEG-DASH ISOFF segmenting muxer
  This is mostly to serve as a reference example on how to segment
  
  the output from the mp4 muxer, capable of writing the segment
  
  list in four different ways :
  
   SegmentTemplate with SegmentTimeline
  
   SegmentTemplate with implicit segments
  
   SegmentList with individual files
  
   SegmentList with one single file per track, and byte ranges
  The muxer is able to serve live content (with optional windowing)
  
  or create a static segmented MPD.
  In advanced cases, users will probably want to do the segmenting
  
  in their own application code.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] Changelog
  • [DBH] configure
  • [DBH] libavformat/Makefile
  • [DBH] libavformat/allformats.c
  • [DBH] libavformat/dashenc.c
  • [DBH] libavformat/version.h

• MPEG-2 AAC frame-by-frame audio decoding

  3 mai 2018, par Jonathan May

  I have individual audio frames encoded in MPEG-2 AAC. Each frame consists of 1024 16 bit PCM samples.

  I notice that each AAC frame is a different size. I assume this is a result of the MPEG-2 AAC compression algorithm and perfectly normal.

  I need a way to decode a single frame and get back the original 1024 PCM samples (with error from lossy compression, that’s fine).

  I couldn’t find information about the MPEG-2 AAC algorithm ANYWHERE online. It’s kinda nuts.

  I’ve been trying a crude work around using a library called pydub, which contains a few methods which use FFMPEG’s AAC decoder. Trying to load the audio frame as an AudioSegment using AAC encoding :

  audioData = BytesIO(frame)
  
  sound = AudioSegment.from_file(audioData, format="aac")

  gives the following error :

  [aac @ 000002d444c1aa00] Estimating duration from bitrate, this may be inaccurate\r\n
  
  Input #0, aac, from 'C:\\Users\\jmk_m\\AppData\\Local\\Temp\\tmpjl3x0xao':\r\n
  
    Duration: 00:00:00.19, bitrate: 23 kb/s\r\n
  
  Stream #0:0: Audio: aac (LC), 22050 Hz, mono, fltp, 23 kb/s\r\nStream mapping:\r\n
  
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))\r\n
  
  Press [q] to stop, [?] for help\r\n
  
  Output #0, wav, to 'C:\\Users\\jmk_m\\AppData\\Local\\Temp\\tmpxmp942e4':\r\n
  
  Metadata:\r\n
  
  ISFT            : Lavf58.10.100\r\n
  
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 352 kb/s\r\n
  
  Metadata:\r\n      encoder         : Lavc58.13.100 pcm_s16le\r\n
  
  [aac @ 000002d444cc7480] Reserved bit set.\r\n
  
  [aac @ 000002d444cc7480] Prediction is not allowed in AAC-LC.\r\n
  
  Error while decoding stream #0:0: Invalid data found when processing input\r\n
  
  [aac @ 000002d444cc7480] Reserved bit set.\r\n
  
  [aac @ 000002d444cc7480] Prediction is not allowed in AAC-LC.\r\n
  
  Error while decoding stream #0:0: Invalid data found when processing input\r\n
  
  [aac @ 000002d444cc7480] Prediction is not allowed in AAC-LC.\r\n
  
  Error while decoding stream #0:0: Invalid data found when processing input\r\n
  
  size=       2kB time=00:00:00.04 bitrate= 366.2kbits/s speed=5.45x    \r\n
  
  video:0kB audio:2kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.808594%\r\n
  
  Conversion failed!\r\n"

  If anyone has any insights as to what may be causing the error, or any alternative approaches, that’d be greatly appreciated !

• How/Where to place the FFMPEG `-legacy_icecast 1` option to only affect one of two Icecast outputs ? (using tee muxer option)

  4 juin 2020, par jzvi12

  I have an FFMPEG command which was able to send an RTMP live feed to 2 different icecast radio stations :

  



  command 1 (in prod ; 1 of 2 failing) :

  



  /home/ubuntu/encoder/bin/ffmpeg -i rtmp://127.0.0.1:1935/app/live -vn -c:a mp3 -ac 2 -b:a 64k -ar 44100 -f tee -map 0 "[f=mp3:onfail=ignore]icecast://source:password@liquidsoap:8000/live|[f=mp3:onfail=ignore]icecast://source:password@fluoz.zeno.fm:80/live"


  



  I would get the following FFMPEG error when running above command 1 :

  



  [tee @ 0x562fffd24bc0] Slave '[f=mp3:onfail=ignore]icecast://source:password@fluoz.zeno.fm:80/live': error opening: End of file
[tee @ 0x562fffd24bc0] Slave muxer #1 failed: End of file, continuing with 1/2 slaves.


  



  It was working fine until these days when one of them (fluoz.zeno.fm:80/live) stopped working and I think Zeno upgraded their Icecast servers from v1 (legacy) to v2. Why ? because when I was doing individual testing to the Zeno icecast server, I realized that it worked fine when I added the -legacy_icecast 1 option

  



  command 2 (testing to Zeno only w/ -legacy_icecast 1 option) :

  



  /home/ubuntu/encoder/bin/ffmpeg -i rtmp://127.0.0.1:1935/app/live -vn -c:a mp3 -ac 2 -b:a 64k -ar 44100 -legacy_icecast 1 -f mp3 icecast://source:password@fluoz.zeno.fm:80/live


  



  QUESTION :

  



  How can I use the command 1 above to still send to both icecast servers but having the -legacy_icecast 1 option to only affect the zeno icecast server and not the liquidsoap icecast server ? I want to avoid running 2 separate instances of ffmpeg at all cost in order to save system resources and bandwidth.

  



  The reason I ask this question is because I tried adding the -legacy_icecast 1 option to the command 1 (that one that sends to both) by placing it right before the ...-f tee... option, but it still didn't work to send to Zeno icecast server which failed (the liquidsoap icecast server kept running fine). It seems to only work fine when I use command 2 (above) when not using the tee muxer option.

  



  Is there a tricky way to place that -legacy_icecast 1 option within the tee muxer option ??