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• avconv : init filtergraphs only after we have a frame on each input

  27 mai 2016, par Anton Khirnov

  avconv : init filtergraphs only after we have a frame on each input
  This makes sure the actual stream parameters are used, which is
  
  important mainly for hardware decoding+filtering cases, which would
  
  previously require various weird workarounds to handle the fact that a
  
  fake software graph has to be constructed, but never used.
  
  This should also improve behaviour in rare cases where
  
  avformat_find_stream_info() does not provide accurate information.
  

  • [DBH] avconv.c
  • [DBH] avconv.h
  • [DBH] avconv_filter.c
  • [DBH] avconv_opt.c
  • [DBH] avconv_qsv.c

• vaapi_encode : Maintain a pool of bitstream output buffers

  5 juin 2016, par Mark Thompson

  vaapi_encode : Maintain a pool of bitstream output buffers
  Previously we would allocate a new one for every frame.  This instead
  
  maintains an AVBufferPool of them to use as-needed.
  Also makes the maximum size of an output buffer adapt to the frame
  
  size - the fixed upper bound was a bit too easy to hit when encoding
  
  large pictures at high quality.
  

  • [DBH] libavcodec/vaapi_encode.c
  • [DBH] libavcodec/vaapi_encode.h

• Recording from online stream and listening to it at the same time (ffmpeg / ffplay)

  28 juin 2016, par Konstantin

  Sometimes I like to record programmes from online radio channels, live or archived streams too. When there is no interesting actual programmes in the radios, I also would like listening to it at the same time while recording. I am using such command lines, which is called from Ruby script - to help parsing radios’ timetables / programme pages and constructing the proper URLs of archived programmes which usually contains some timecode, such as 20160616_083000.mp3, etc.
  So my command line to call from Ruby script looks like :

      programmes.each{|datepart,programme_length| 
  
        cmd=%Q{ffmpeg -y -i http://example.com/stream/#{datepart}.mp3 -t #{programme_length} -c:a libmp3lame -b:a 160k "#{fname}" -c copy -t #{programme_length} -f mp3 -f rtp rtp://127.0.0.1:8888}
  
        system cmd
  
  }

  It resides in a loop to record the previously parsed and selected programmes. Of cource the programmes are recorded properly and at the same time ffmpeg streams it as an mp3 rtp stream as well on localhost at the given port. In another terminal window I connect to the streamed data with one-liner as follows :

  while true; do ffplay -i rtp://127.0.0.1:8888 -autoexit; done

  I am using the -autoexit switch which should be stop playing the stream when it is ended and the "while" loop should be connect again to the new stream which is served by the programme recording "each" loop. Unfortunately it keeps playing after the end, and doesn’t initiate a new connection to the newly started stream. How to use ffplay properly to stop playing after rtp stream is ended and let it connect again to the new stream ?