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• FFMPEG : Microphone capturing too much noise while using ffmpeg command

  29 octobre 2020, par Rupesh J

  I am listening audio from source IP address and trying to encode it into speex format and again sending it to the destination IP address using ffmpeg.

  


  My ffmpeg command is :

  


  ffmpeg -protocol_whitelist file,rtp,udp -i temp.sdp -c:a libspeex -f rtp rtp://:<port>&#xA;</port>

  


  SDP file content is(temp.sdp) :

  


  v=0 &#xA;c=IN IP4 &#xA;t=0 0&#xA;m=audio <port> RTP/AVP 98&#xA;a=rtpmap:98 L16/8000&#xA;</port>

  


  Issue : Whenever I am trying to run this command, I am getting too much background noise on speaker.
I could hear music(not clearly), but not human voice.

  


  Also, I have tried with highpass and lowpass filters are as follows :

  


  ffmpeg -protocol_whitelist file,rtp,udp -i temp.sdp -af "highpass=f=200, lowpass=f=3000" -c:a &#xA;    libspeex -f rtp rtp://:<port>&#xA;</port>

  


• 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.

  


• Normalizing audio in ffmpeg - how ?

  11 novembre 2020, par Betty Crokker

  I'm creating one of those "Brady Bunch" videos for a choir using a C# application I'm writing that uses ffmpeg for all the heavy lifting, and for the most part it's working great but I'm having trouble getting the audio levels just right.

  


  What I'm doing right now, is first "normalizing" the audio from the individual singers like this :

  


  

  • Extract audio into a WAV file using ffmpeg

  


  • Load the WAV file into my application using NAudio

  


  • Find the maximum 16-bit value

  


  • When I create the merged video, specify a volume for this stream that boosts the maximum value to 32767

  


  


  So, for example, if I have 3 streams : stream A's maximum audio is 32767 already, stream B's maximum audio is 32000, and stream C's maximum audio is 16000, then when I merge these videos I will specify

  


  [0:a]volume=1.0,aresample=async=1:first_pts=0[aud0]
[1:a]volume=1.02,aresample=async=1:first_pts=0[aud1]
[2:a]volume=2.05,aresample=async=1:first_pts=0[aud2]
[aud0][aud1][aud2]amix=inputs=3[a]


  


  (I have an additional "volume tweak" that lets me adjust the volume level of individual singers as necessary, but we can ignore that for this question)

  


  I am reading the ffmpeg wiki on Audio Volume Manipulation, and I will implement that next, but I don't know what to do with the output it generates. It looks like I'm going to get mean and max volume levels in dB and while I understand decibels in a "yeah, I learned about those in college 30 years ago" kind of way, I don't know how to use those values to normalize the audio of my input videos.

  


  The problem is, in the ffmpeg output video, the audio level is quite low. If I do the same process of extracting the audio and looking at the WAV file in the merged video that ffmpeg generated, the maximum value is only 4904.

  


  How do I implement an algorithm that automatically sets the output volume to a "reasonable" level ? I realize I can simply add a manual volume filter and have the human set the level, but that's going to be a lot of back & forth of generating the merged video, listening to it, adjusting the level, merging again, etc. I want a way where my application figures out an appropriate output volume (possibly with human adjustment allowed).

  


  EDIT

  


  Asking ffmpeg to determine the mean and max volume of each clip does provide mean and max volume in dB, and I can then use those values to scale each input clip :

  


  [0:a]volume=3.40dB,aresample=async=1:first_pts=0[aud0]
[1:a]volume=3.90dB,aresample=async=1:first_pts=0[aud1]
[2:a]volume=4.40dB,aresample=async=1:first_pts=0[aud2]
[3:a]volume=-0.00dB,aresample=async=1:first_pts=0[aud3]


  


  But my final video is still strangely quiet. For now, I've added a manually-entered volume factor that gets applied at the very end :

  


  [aud0][aud1][aud2]amix=inputs=3[a]
[a]volume=volume=3.00[b]


  


  So my question is, in effect, how do I determine algorithmically what this final volume factor needs to be ?

  


  MORE EDIT

  


  There's something deeper going on here, I just set the volume filter to 100 and the output is only slightly louder. Here are my filters, and the relevant portions of the command line :

  


  color=size=1920x1080:c=0x0000FF [base];
[0:v] scale=576x324 [clip0];
[0:a]volume=1.48,aresample=async=1:first_pts=0[aud0];
[1:v] crop=808:1022:202:276,scale=384x486 [clip1];
[1:a]volume=1.57,aresample=async=1:first_pts=0[aud1];
[2:v] crop=1160:1010:428:70,scale=558x486 [clip2];
[2:a]volume=1.66,aresample=async=1:first_pts=0[aud2];
[3:v] crop=1326:1080:180:0,scale=576x469 [clip3];
[3:a]volume=1.70,aresample=async=1:first_pts=0[aud3];
[4:a]volume=0.20,aresample=async=1:first_pts=0[aud4];
[5:a]volume=0.73,aresample=async=1:first_pts=0[aud5];
[6:v] crop=1326:1080:276:0,scale=576x469 [clip4];
[6:a]volume=1.51,aresample=async=1:first_pts=0[aud6];
[base][clip0] overlay=shortest=1:x=32:y=158 [tmp0];
[tmp0][clip1] overlay=shortest=1:x=768:y=27 [tmp1];
[tmp1][clip2] overlay=shortest=1:x=1321:y=27 [tmp2];
[tmp2][clip3] overlay=shortest=1:x=32:y=625 [tmp3];
[tmp3][clip4] overlay=shortest=1:x=672:y=625 [tmp4];
[aud0][aud1][aud2][aud3][aud4][aud5][aud6]amix=inputs=7[a];
[a]adelay=delays=200:all=1[b];
[b]volume=volume=100.00[c];
[c]asplit[a1][a2];

ffmpeg -y ....
   -map "[tmp4]" -map "[a1]" -c:v libx264 "D:\voutput.mp4" 
   -map "[a2]" "D:\aoutput.mp3""


  


  When I do this, the audio I want is louder (loud enough to clip and get distorted), but definitely not 100x louder.