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• How to quit pexpect launched ffmpeg with key q pressed

  25 février 2014, par Shuman

  i used pexpect to call ffmpeg which is a lengthy process. it took half an hour, how can i detect user has pressed q key to stop it ? just like when you press q when using ffmpeg command line tool

  the ffmpeg command line is
  ffmpeg -y -i url -c copy -absf aac_adtstoasc out.mp4

  the last line of ffmpeg output is

  ...
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
    Stream #0:2 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=   84 fps= 77 q=-1.0 Lsize=  184626kB time=00:00:06.96 bitrate=217120.3kbits/s

  the code i have now is

  reo = re.compile("""\S+\s+(?P\d+)  # frame
  
                       \s\S+\s+(?P<fps>\d+)           # fps
  
                       \sq=(?P<q>\S+)                    # q
  
                       \s\S+\s+(?P<size>\S+)          # size
  
                       \stime=(?P<time>\S+)           # time
  
                       \sbitrate=(?P<bitrate>[\d\.]+) # bitrate
  
                       """, re.X)
  
  
  
  durationReo = (&#39;(?&lt;=Duration:\s)\S+(?=,)&#39;)
  
  
  
  cpl = thread.compile_pattern_list([
  
      pexpect.EOF,
  
      reo,
  
      durationReo
  
  ])
  
  
  
  while True:
  
      i = thread.expect_list(cpl, timeout=None)
  
      if i == 0: # EOF
  
          print "the sub process exited"
  
          break
  
      elif i == 1:
  
          frame_number = thread.match.group(0)
  
          print frame_number
  
          print reo.search(frame_number).groups()
  
          # thread.close
  
      elif i == 2:
  
          durationLine = thread.match.group(0)
  
          print &#39;Duration:&#39;, durationLine
  
          # print "something :",thread.match.group(1)
  
          pass
  
  </bitrate></time></size></q></fps>

  with this code i can already get the frame info and duration info, the ultimate goal is to create a textual progress bar with another python progressbar module. but with the ability to send the 'q' pressed signal to ffmpeg child process.

• Detecting the value scale of statistics returned from ffprobe

  15 septembre 2023, par Farski

  I'm using ffprobe to detect max and min levels for various audio files. An example of the command I'm using is :

  


  ffprobe -v error -f lavfi -i amovie=my_song.mp3,asetnsamples=n=4410,astats=metadata=1:reset=1 -show_entries frame_tags=lavfi.astats.Overall.Max_level,lavfi.astats.Overall.Min_level -of json

  


  The max/min level values returned use different scales, depending on the format of the input file.

  


  For example, an MP3 file may return fractional values from -1.0 to 1.0 representing a percent of maximum level. A signed 16 bit WAV file returns values in the range -32,768 to 32767. A signed 32 bit FLAC file uses the range -2,147,483,648 to 2,147,483,647. In these cases, the bit size of the values matches the bit depth of the audio file.

  


  In other cases, such as a signed 8 bit WAV file, the results are returned using a scale that does not match the input file, such as 16 bit scale (-32,768 to 32767).

  


  I'm trying to determine if there's anyway to detect which format or scale ffprobe is using when the results are returned, besides trying to do it heuristically. I haven't been able to find any other value that gets returned which specifically reflects the number system being used to generate these levels values. sample_fmt does, in some cases, match, but in cases such as a s8 WAV file, sample_fmt would return s8, which does not match the number format of the returned levels values (s16).

  


  If it's not possible to request this information from ffprobe JIT, is there anywhere in the code base that would describe how it determines which scale to use ?

  


• How can i decode an mp3 and encode it as aac with ezstream

  8 avril 2015, par Roberto Arosemena

  This is my current ezstream config

  <ezstream>
  
     <url>http://localhost:8000/test</url>
  
     <sourcepassword>password</sourcepassword>
  
     <format>MP3</format>
  
     <filename>playlist.m3u</filename>
  
     <reencode>
  
        <enable>1</enable>
  
        <encdec>
  
           <format>MP3</format>
  
           <match>.mp3</match>
  
           <decode>madplay -b 16 -R 44100 -S -o raw:- "@T@"</decode>
  
           <encode>lame --preset cbr 32 -r -s 44.1 --bitwidth 16 - -</encode>
  
        </encdec>
  
     </reencode>
  
  </ezstream>

  It’s mounting to an icecast server, its decoding and encoding mp3 to a lower bitrate, I’m trying to encode it to aac instead of mp3 in hopes that the quality improves as i heard that aac is better than mp3 for lower bitrates.

  What i would like to know is if i can use an aac encoder such as FFmpeg instead of the lame mp3 encoder and get an aac to the end user instead of mp3, if so what parameters should i pass to FFmpeg so that it works with my current config.