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• aaccoder : tweak PNS implementation further

  9 septembre 2015, par Rostislav Pehlivanov

  aaccoder : tweak PNS implementation further
  This commit changes a few things about the noise substitution
  
  logic :
  
   - Brings back the quantization factor (reduced to 3) during
  
     scalefactor index calculations.
  
   - Rejects any zeroed bands. They should be inaudiable and it’s
  
     a waste transmitting the scalefactor indices for these.
  
   - Uses swb_offsets instead of incrementing a ’start’ with every
  
     window group size.
  
   - Rejects all PNS during short windows.
  
  Overall improves quality. There was a plan to use the lfg system
  
  to create the random numbers instead of using whatever the decoder
  
  uses but for now this works fine. Entropy is far from important here.
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] libavcodec/aaccoder.c
  • [DH] tests/fate/aac.mak

• Is there a way to use ffmpeg audio filters to automatically synchronize 2 streams with similar content

  29 mai 2015, par user3741412

  I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a ’line in’ jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.

  The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.

  So I can do this today :

  • Take the good sound and the camera captured sound and load each into
    audacity and pretty quickly use the timeshift toot to perfectly fit
    the good audio to the questionable audio from the HDMI capture and
    cut the good audio to the exact size of the video. Then I can use
    ffmpeg or other video editing software to replace the questionable
    audio with the better audio.

  But while somewhat quick and easy, it always carries with it a bit of human error and time. I’d like to automate this if possible as this process is repeated at least weekly throughout the year.

  Does anyone have a suggestion if any of these ideas have merit or could suggest another approach ?

  1. I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn’t see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats ?

     • if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now— I’ve never tried either of these filters yet— just trying to optimize against blind alleys by asking for advice up front.

  2. If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I’d take it, if it turns out reasonably reliable

  3. Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum ? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible— but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.

  4. any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI ?

  Thanks in advance for any pointers or suggestinons !

• Evolution #4391 : Squelettes de la dist : améliorer le markup et passer à BEM

  18 octobre 2019, par nicod _

  RastaPopoulos ♥ a écrit :

  @nico le "plan" est bien un modifier chez tcharlss, dans BEM il n’y a plus "—" depuis pas mal de temps dans la doc officielle. La version en cours c’est DEUX "_" = un sous-élément, UN "_" = un modifier. :)
  Les tirets ne sont plus utilisés que pour séparer les termes multi-mots (comme le disait tcharlss plus haut, à la place du camelCase).

  N’importe quoi, ça devient complètement illisible...

  Groumpfff... tant pis, je ferai du BEM de vieux con dans mon coin.