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• Make sound settings created by ffmpeg apply to entire Windows applications [closed]

  7 septembre 2023, par rabeh koo

  The recording sound of my microphone is weak, so I resorted to using ffmpeg to add a volume increase effect and some other effects. I created a simple batch file and recorded the sound with it, and it works very well.
This is the batch file code :

  


  ffmpeg -f dshow -i audio="Microphone (USBAudio2.0)" -channels 1 -af " stereotools=level_in=2 , afftdn=nf=-25 , volume=10 " Output.wav


  


  

  • But the problem occurs when I want to make calls or use the microphone in Windows applications, the recording is very weak. Is there a way to make the ffmpeg effects that are in the batch file apply to all system applications, and how can that be done ?

  


  


  I searched and found that I have to create a virtual microphone device with some programs and make it the main microphone for the system. I tried these programs, but there was no option to add ffmpeg codes or pass the audio through the batch file that I created earlier.

  


• ffmpeg configuration difficulty with filter_complex and hls

  4 février 2020, par akc42

  I am trying to set up ffmpeg so that it will record from a microphone and encode the results at the same time into a .flac file for later syncing up with some video I will be making.

  The microphone is plugged into a raspberry pi (4B) and I am currently trying it with a blue yeti mic, but I think I can do the same with a focusrite scarlett 2i2 plugged in instead. However I was puzzling about how to start the server recording and decided I could do it from a web browser if I made a simple nodejs server that spawned ffmpeg as a child process.

  But then I was inspired by this sample ffmpeg command which displays (on my desktop with an graphical interface) a volume meter

  ffmpeg -hide_banner -i 'http://distribution.bbb3d.renderfarming.net/video/mp4/bbb_sunflower_1080p_30fps_normal.mp4' -filter_complex "showvolume=rate=25:f=0.95:o=v:m=p:dm=3:h=80:w=480:ds=log:s=2" -c:v libx264 -c:a aac -f mpegts - | ffplay -window_title "Peak Volume" -i -

  What if I could stream the video produced by the showvolume filter to the web browser that I am using to control the ffmpeg process (NOTE I don’t want to send the audio with this). So I tried to read up on hls (since the control device will be an ipad - in fact that is what I will record the video on), and came up with this command

  ffmpeg -hide_banner -f alsa -ac 2 -ar 48k -i hw:CARD=Microphone -filter_complex "asplit=2[main][vol],[vol]showvolume=rate=25:f=0.95:o=v:m=p:dm=3:h=80:w=480:ds=log:s=2[vid]" -map [main] -c:a:0 flac recordings/session_$(date +%a_%d_%b_%Y___%H_%M_%S).flac -map [vid] -preset veryfast -g 25 -an -sc_threshold 0 -c:v:1 libx264 -b:v:1 2000k -maxrate:v:1 2200k -bufsize:v:3000k -f hls -hls_time 4 -hls_flags independent_segments delete_segments -strftime 1 -hls_segment_filename recordings/volume-%Y%m%d-%s.ts recordings/volume.m3u8

  The problem is I am finding the documentation a bit opaque as to what happens once I have generated two streams - the main audio and a video stream, and this command throws both a warning and an error :-

  The warning is Guessed Channel Layout for Input Stream #0.0 : stereo

  and the error is

  [NULL @ 0x1baa130] Unable to find a suitable output format for 'hls'
  
  hls: Invalid argument

  What I am trying to do is set up stream labels [main] and [vol] as I split the incoming audio into two parts, then I pass [vol] through the "showvolume" filter and end up with stream [vid].

  I think I need to then use -map to specify encoding the [main] stream down to flac and writing it out to file (The file exists after I run the command although they have zero length), and use another -map to pass through to the -f hls section. But I think I have something wrong at this stage.

  Can someone help me get this command right.

• HTML5 icecast stream and pausing stream to add advert

  2 avril 2019, par RussellHarrower

  So I have not found a good solution to add audio advertising to our radio station for users who are not in the stations main area (western Australia).

  So I was wondering is there away that anyone knows of, that would allow a code be it ffmpeg or java/JavaScript to listen for the song that is currently playing (it maybe delayed by 30seconds) and after the song ends start an audio ad from a mongodb database.

  I know how to do the mongodb database part.

  What I need to figure out is how to detect that the song is finishing or other option is to store the stations adverts in the db as raw data or even better we could air a 0.1sec file that is just blank and that filename tells the script to lower audio player ones volume and boost audio two volume.

  When station ads end we play another filename with same length and reverse the process.

  Ok sorry if that’s me thinking out loud but if you have any ideas that would be great.

  The only issue with above is TuneIn app won’t work with an IFrame we need to supply them an audio url.

  So if anyone know how to do this in ffmpeg and only require 1 url which can detect users location that be better solution.

  All ideas welcomed and yes I googled and Bing searched however I don’t have 30,000$ for hardware companies are selling. There has to be an open source solution.