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• Using an actual audio recording to filter out noise from a video

  9 mars 2021, par user2751530

  I use my laptop (Ubuntu 18.04 LTS derivative on a Dell XPS13) for recording videos (these are just narrated presentations) using OBS. After a presentation is done (.flv format), I process it using ffmpeg using filters that try to reduce background noise, reduce the size of the video, change encoding to .mp4, insert a watermark, etc. Over several months, this system has worked well.

  


  However, my laptop is now beginning to show its age (it is 4 years old). That means that the fan becomes loud - loud enough to notice in a recording, not loud enough to notice when you are working. So, even after filtering for low frequency in ffmpeg, there are clicking and other type of sounds that are left in the video. I am a scientist, though not an audio/video expert. So, I was thinking - is it possible for me to simply record the noise coming out of my machine when I am not presenting, and then use that recording to filter out the noise that my machine makes during the presentation ?

  


  Blanket approaches like filtering out certain ranges of the audio spectrum, etc. are unlikely to work, as the power spectrum of the noise likely has many peaks, and these are likely to extend into human voice range as well (I can hear them). Further, this is a moving target - the laptop is aging and in any case, the amount and type of noise it makes depends on the load and how long it has been on. Algorithm :

  


  

  1. Record actual computer noise (with the added bonus of background noise) while I am not recording. Ideally, just before starting to record the presentation. This could take the form of a 1-2 minute audio sample.

  


  2. Record the presentation on OBS.

  


  3. Use 1 as a filter to get rid of noise in 2. I imagine it would involve doing a Fourier analysis of 1, and then removing those peaks from the spectrum of 2 at each time epoch.

  


  


  I have looked into sox, which is what people somewhat flippantly point you to without giving any details. I do not know how to separate out audio channels from a video and then interleave them back together (not an expert on the software here). Other than RTFM, is there any helpful advice anyone could offer ? I have searched, but have not been able to find a HOWTO. I expect that that is probably the fault of my search since I refuse to believe that this is a new idea - it is a standard method used in many fields to get rid of noise, including astronomy.

  


• avformat/pcm : factorize and improve determining the default packet size

  2 mars 2024, par Marton Balint

  avformat/pcm : factorize and improve determining the default packet size
   Remove the 1024 cap on the number of samples, for high sample rate audio it
  
    was suboptimal, calculate the low neighbour power of two for the number of
  
    samples (audio blocks) instead.
  
   Make the function work correctly also for non-pcm codecs by using the stream
  
    bitrate to estimate the target packet size. A previous version of this patch
  
    used av_get_audio_frame_duration2() the estimate the desired packet size, but
  
    for some codecs that returns the duration of a single audio frame regardless
  
    of frame_bytes.
  
   Fallback to 4096/block_align*block_align if bitrate is not available.
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] libavformat/pcm.c
  • [DH] libavformat/pcm.h
  • [DH] tests/ref/seek/lavf-al
  • [DH] tests/ref/seek/lavf-ul

• Real-Time Buffer Too Full (FFMPEG)

  25 janvier 2018, par Nimble

  So I’ve been having this issue with ffmpeg, it has been a journey getting the hardware and command to actually do what I want, but I still have one problem.

  Sometimes when I’m recording I just start dropping frames like crazy, this can be after an hour of recording or even ten hours in... Everything will be working fine and then suddenly I’ll start dropping frames due to "real-time buffer too full or near too full". This happens regardless of how low I put the bitrate, and the buffer size is high as it will allow, eventually I’ll just start dropping frames. Almost seems like it could be like a power saving feature kicking in but it’s too inconsistent it seems. Like I said sometimes I can go 10 hours without having this issue.

  Any ideas ?

  Here is my block of code :

  ffmpeg -guess_layout_max 0 -y -f dshow -video_size 3440x1440 -rtbufsize 2147.48M -pixel_format nv12 -framerate 200 ^
  
  -i video="Video (00 Pro Capture HDMI 4K+)":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" -map 0:0,0:1 -map 0:1 ^
  
  -preset: llhp -codec:v h264_nvenc -pix_fmt nv12 -b:v 250M -maxrate:v 250M -minrate:v 250M -bufsize:v 250M -b:a 320k ^
  
  -ac 2 -r 100 -async 1 -vsync 1 -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\PC%02d.mp4 ^
  
  -guess_layout_max 0 -f dshow -rtbufsize 2000M -i audio="Analog (3+4) (RME Fireface UC)" -map 1:0 -b:a 320k -ac 2 ^
  
  -af "adelay=200|200" -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\Voices\Theirs\TPC%02d.wav ^
  
  -guess_layout_max 0 -f dshow -rtbufsize 2000M -i audio="Analog (5+6) (RME Fireface UC)" -map 2:0 -b:a 320k -ac 2 ^
  
  -af "adelay=825|825" -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\Voices\Mine\MPC%02d.wav

  Here is the error, it repeated around 300 times before locking up ffmpeg forcing my to quit before starting the recording again :

  [dshow @ 0000019a596bdcc0] real-time buffer [Video (00 Pro Capture HDMI 4K+)] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!