Sur d’autres sites (11765)

• waveforms arent as smooth as they should be

  25 mars 2019, par GROVER.

  So I’ve been trying to create a little clone of SoundClouds ’peaks’ waveform. I have noticed that their waveforms are a lot more streamlined
  compared to my sudden shifts in ’loudness’. The dynamic range between peaks never seems to be too steep, and they always represent the perceived loudness of each part of the track pretty well.

  ---

  Here’s an example :

  

  Notice how all the ’drops’ are fairly distinguishable to the ’breakdowns’ in their waveform, but mine is all over the place (apart from the last drop and breakdown, which are kind of similar). There are some minor similarities, but the ’jaggedness’ is still very prominent even in those areas.

  I’m using wav2json as a peaks converter (which is run through the command line and programmed in C++). This is is an example of how I use it :

  /*
  
  *    --channels: mids and min
  
  *    --db-min (minimum level in dB to capture): -35dB
  
  *    --db-max (pretty self explanatory): 6dB
  
  *    -d: use logarithmic instead of linear scale
  
  *    -s (number of peaks to generate): 1800
  
  *    -o (output file): outputfile.json
  
  *    -p (precision of floats): 0
  
  *    -n: no header 
  
  */
  
  exec("wav2json inputfile.wav -s 1800 --channels mid min -d --db-min -35 --db-max 6 -p 0 -o outputfile.json -n");
  
  
  
  $fp     = fopen($tmpOutput, "r");
  
  $json   = fread($fp, filesize($tmpOutput));
  
  // get mids and min from the generated peaks file
  
  $mid    = json_decode($json, true)["mid"];
  
  $min    = json_decode($json, true)["min"];
  
  fclose($fp);
  
  unlink($tmpOutput);
  
  
  
  /* 
  
  *  from here I just combine each mid and min value together and divide by two
  
  *
  
  *  then I normalise all the peaks (instead of each value being between -0.293 to 
  
  *  1.766(just as an example), it is between 0 and 100)
  
  */

  What I’m trying to figure out - and have been trying to for the last few months - is how to get each peak more streamlined and to have the dynamic range of each one look how it actually sounds.

  ---

  What I have tried :

  • ffmpeg eqing
  • actually eqing the highs and lows in a daw and then comparing waveforms
  • using various parameters for wav2json (db min and max, linear etc.)
  • using various compressors and multiband compressors on the track

  All help is appreciated,
  Cheers.

• ffmpeg being inprecise when trimming mp3 files

  12 janvier 2019, par cactus

  I want to use ffmpeg to trim some mp3s without re-encoding. The command I used was
  ffmpeg -i "inputfile.mp3" -t 00:00:12.414 -c copy out.mp3
  However, out.mp3 has a length of 12.460s, and when I load the file in Audacity I can see that it was cut at the wrong spot, and not at 12.414s.

  Why is this ? I googled a bit and tried some other commands like ffmpeg -i "inputfile.mp3" -ss 0 -to 00:00:12.414 -c copy out.mp3 (which interestingly results in a different length of 12.434s) but could never get the milliseconds to be cut right.

  

  

  PS. I wasn’t sure whether SO was the right place to ask since it isn’t technically programming related, however most of the stuff I found on ffmpeg for trimming audio files were stackoverflow questions, e. g. ffmpeg trimming videos with millisecond precision

• avcodec/mpeg : Initialize quarter_sample parameter from previous thread.

  13 décembre 2018, par Andriy Gelman

  avcodec/mpeg : Initialize quarter_sample parameter from previous thread.
  Fixes #7410.
  
  The value of sub-pixel precision for me/mc can change during an Intra frame. In multi-threaded decoding this change is not propagated to other frame threads causing decoding artifacts. This patch initializes the sub-pixel precision parameter from previous thread, which fixes the issue.
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/mpegvideo.c