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• avcodec/siren : Increase noise category 5 and 6

  24 décembre 2020, par Michael Niedermayer

  avcodec/siren : Increase noise category 5 and 6
  The entry read is not used in subsequent computation, thus its
  
  value is not important.
  Fixes : out of array read
  
  Fixes : 28578/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SIREN_fuzzer-6332019122503680
  Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/siren.c

• 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.

  


• modprobe snd-dummy on linux server, ffmpeg -f alsa is recording noise

  20 mars 2021, par booksword

  Some people say that the parameter fake_buffer can remove the noise, but I tried it and there is still noise

  


  


  modprobe snd-dummy fake_buffer=0

  


  


  #modprobe snd-dummy fake_buffer=0 
# lsmod |grep snd
snd_dummy              24576  0
snd_pcm_oss            61440  0
snd_mixer_oss          28672  1 snd_pcm_oss
snd_pcm               118784  2 snd_pcm_oss,snd_dummy
snd_timer              36864  1 snd_pcm
snd                    94208  5 snd_timer,snd_pcm_oss,snd_pcm,snd_dummy,snd_mixer_oss
soundcore              16384  1 snd
# modinfo snd_dummy
filename:       /lib/modules/4.14.129-bbrplus/kernel/sound/drivers/snd-dummy.ko
license:        GPL
description:    Dummy soundcard (/dev/null)
author:         Jaroslav Kysela <perex@perex.cz>
depends:        snd-pcm,snd
retpoline:      Y
intree:         Y
name:           snd_dummy
vermagic:       4.14.129-bbrplus SMP mod_unload modversions 
parm:           index:Index value for dummy soundcard. (array of int)
parm:           id:ID string for dummy soundcard. (array of charp)
parm:           enable:Enable this dummy soundcard. (array of bool)
parm:           model:Soundcard model. (array of charp)
parm:           pcm_devs:PCM devices # (0-4) for dummy driver. (array of int)
parm:           pcm_substreams:PCM substreams # (1-128) for dummy driver. (array of int)
parm:           fake_buffer:Fake buffer allocations. (bool)
parm:           hrtimer:Use hrtimer as the timer source. (bool)

#aplay -L
null
    Discard all samples (playback) or generate zero samples (capture)
pulse
    PulseAudio Sound Server
dummy
default:CARD=Dummy
    Dummy, Dummy PCM
    Default Audio Device
sysdefault:CARD=Dummy
    Dummy, Dummy PCM
    Default Audio Device
dmix:CARD=Dummy,DEV=0
    Dummy, Dummy PCM
    Direct sample mixing device
dsnoop:CARD=Dummy,DEV=0
    Dummy, Dummy PCM
    Direct sample snooping device
hw:CARD=Dummy,DEV=0
    Dummy, Dummy PCM
    Direct hardware device without any conversions
plughw:CARD=Dummy,DEV=0
    Dummy, Dummy PCM
    Hardware device with all software conversions


# ffmpeg -f alsa  -i hw:0 -t 5  /dev/shm/out.mp3 -y
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
  configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'hw:0':
  Duration: N/A, start: 1616225206.353453, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to '/dev/shm/out.mp3':
  Metadata:
    TSSE            : Lavf57.83.100
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc57.107.100 libmp3lame
size=      79kB time=00:00:05.01 bitrate= 129.0kbits/s speed=   1x     
video:0kB audio:79kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.293899%




  


  i use alsamixer to set all Volume to 0 ,and there is still noise
I searched in google for a long time and did not find other people have this problem , where am I doing it wrong