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• AAC encoder : fix possible assertion failure in PNS

  22 décembre 2015, par Claudio Freire

  AAC encoder : fix possible assertion failure in PNS
  Fix possible SF delta violation that would cause an
  
  eventual assertion failure in some corner cases (esp
  
  on very low bitrates) when marking bands for PNS due
  
  to misuse of the sf_delta utilities
  

  • [DH] libavcodec/aaccoder.c

• Revision 4210 : mini bug

  18 octobre 2010, par kent1 — Log

  mini bug

• Gaps when recording using MediaRecorder API(audio/webm opus)

  9 août 2018, par Jack Juiceson

  ----- UPDATE HAS BEEN ADDED BELOW -----

  I have an issue with MediaRecorder API (https://www.w3.org/TR/mediastream-recording/#mediarecorder-api).

  I’m using it to record the speech from the web page(Chrome was used in this case) and save it as chunks.
  I need to be able to play it while and after it is recorded, so it’s important to keep those chunks.

  Here is the code which is recording data :

  navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(function(stream) {
 recorder = new MediaRecorder(stream, { mimeType: 'audio/webm; codecs="opus"' })
 recorder.ondataavailable = function(e) {
   // Read blob from `e.data`, decode64 and send to sever;
 }
 recorder.start(1000)
})

  The issue is that the WebM file which I get when I concatenate all the parts is corrupted(rarely) !. I can play it as WebM, but when I try to convert it(ffmpeg) to something else, it gives me a file with shifted timings.

  For example. I’m trying to convert a file which has duration 00:36:27.78 to wav, but I get a file with duration 00:36:26.04, which is 1.74s less.

  At the beginning of file - the audio is the same, but after about 10min WebM file plays with a small delay.

  After some research, I found out that it also does not play correctly with the browser’s MediaSource API, which I use for playing the chunks. I tried 2 ways of playing those chunks :

  In a case when I just merge all the parts into a single blob - it works fine.
  In case when I add them via the sourceBuffer object, it has some gaps (i can see them by inspecting buffered property).
  697.196 - 697.528 ( 330ms)
  996.198 - 996.754 ( 550ms)
  1597.16 - 1597.531 ( 370ms)
  1896.893 - 1897.183 ( 290ms)

  Those gaps are 1.55s in total and they are exactly in the places where the desync between wav & webm files start. Unfortunately, the file where it is reproducible cannot be shared because it’s customer’s private data and I was not able to reproduce such issue on different media yet.

  What can be the cause for such an issue ?

  ----- UPDATE -----
  I was able to reproduce the issue on https://jsfiddle.net/96uj34nf/4/

  In order to see the problem, click on the "Print buffer zones" button and it will display time ranges. You can see that there are two gaps :
  0 - 136.349, 141.388 - 195.439, 197.57 - 198.589

  1. 136.349 - 141.388
  2. 195.439 - 197.57

  So, as you can see there are 5 and 2 second gaps. Would be happy if someone could shed some light on why it is happening or how to avoid this issue.

  Thank you