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• lavu : Add AVSphericalMapping type and frame side data

  11 novembre 2016, par Vittorio Giovara

  lavu : Add AVSphericalMapping type and frame side data
  While no decoder currently exports spherical information, this type
  
  represents a frame property that has to be passed through from container
  
  to frames.
  Signed-off-by : Vittorio Giovara <vittorio.giovara@gmail.com>
  

  • [DH] doc/APIchanges
  • [DH] libavutil/Makefile
  • [DH] libavutil/avutil.h
  • [DH] libavutil/frame.h
  • [DH] libavutil/spherical.c
  • [DH] libavutil/spherical.h
  • [DH] libavutil/version.h

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

  25 mars 2019, 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

• 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