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• 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 !

• avformat/matroskaenc : Don't waste bytes on BlockGroup length fields

  16 janvier 2022, par Andreas Rheinhardt

  avformat/matroskaenc : Don't waste bytes on BlockGroup length fields
  This commit uses the new EbmlWriter API to write the length fields
  
  of the BlockGroup and its descendants that are themselves Master
  
  elements (namely BlockAdditions and BlockMore) on the least amount of
  
  bytes.
  This fixes regressions introduced when the special code for writing
  
  general subtitles was removed. Accordingly, the binsub-mksenc and
  
  matroska-zero-length-block FATE-tests have now been reverted back
  
  to their old state again ; the advantages of this approach are evident
  
  with the matroska-vp8-alpha-remux test which up until now wrote
  
  all the length fields of all BlockGroups, BlockAdditions and BlockMore
  
  on eight bytes.
  Using the EbmlWriter API also allowed to improve locality in
  
  mkv_write_block() : E.g. both DiscardPadding as well as the
  
  BlockAdditional side-data are now directly used to add elements
  
  to the writer whereas the earlier code had to first check
  
  for whether a BlockGroup should be used and then check again
  
  (after the place where a BlockGroup would be opened if one were
  
  used) for whether there is DiscardPadding or BlockAdditional
  
  side-data to write.
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavformat/matroskaenc.c
  • [DH] tests/ref/fate/binsub-mksenc
  • [DH] tests/ref/fate/matroska-vp8-alpha-remux
  • [DH] tests/ref/fate/matroska-zero-length-block
  • [DH] tests/ref/fate/webm-dash-chapters

• How to stop ffmpeg when there's no incoming rtmp stream

  5 juillet 2016, par M. Irich

  I use ffmpeg together with nginx-rtmp.
  The thing is ffmpeg doesn’t finish the process when the stream’s finished

  I use the following command :

  ffmpeg  -i 'rtmp://localhost:443/live/test' -loglevel debug  -c:a libfdk_aac -b:a 192k -c:v libx264 -profile baseline -preset superfast -tune zerolatency -b:v 2500k -maxrate 4500k -minrate 1500k -bufsize 9000k -keyint_min 15 -g 15 -f dash -use_timeline 1 -use_template 1 -min_seg_duration 5000 -y /tmp/dash/test/test.mpd

  but even the stream’s not running ffmpeg still can’t finish the process and is waiting for the rtmp stream

  Successfully parsed a group of options.
  
  Opening an input file: rtmp://localhost:443/live/test.
  
  [rtmp @ 0x2ba2160] No default whitelist set
  
  [tcp @ 0x2ba2720] No default whitelist set
  
  [rtmp @ 0x2ba2160] Handshaking...
  
  [rtmp @ 0x2ba2160] Type answer 3
  
  [rtmp @ 0x2ba2160] Server version 13.14.10.13
  
  [rtmp @ 0x2ba2160] Proto = rtmp, path = /live/test, app = live, fname = test
  
  [rtmp @ 0x2ba2160] Server bandwidth = 5000000
  
  [rtmp @ 0x2ba2160] Client bandwidth = 5000000
  
  [rtmp @ 0x2ba2160] New incoming chunk size = 4096
  
  [rtmp @ 0x2ba2160] Creating stream...
  
  [rtmp @ 0x2ba2160] Sending play command for 'test'

  Is it possible to limit the latency time to several seconds ?

  Sorry for any possible mistakes - English’s not my native language.