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• FFMPEG - Struggling to find correct input audio codec parameters on macOS

  20 septembre 2024, par Xavi

  I am trying to read my external stereo microphone with ffmpeg within my Qt Windows+macOs application, but I am struggling to obtain consistent correct input codec parameters on macOs. My findings and suspicions so far :

  


  The code I'm using in macOs is the following, where everything returns a successful return code :

  


   avdevice_register_all();
 
 //macOs only, the same code in windows looks for "dshow" 
 const AVInputFormat *inputFormat = av_find_input_format("avfoundation");
 
 AVFormatContext* inputFormatContext;
 avformat_open_input(&inputFormatContext, inputDevice, inputFormat, NULL);

 avformat_find_stream_info(inputFormatContext, NULL);
 
 //... allocate the codec context for the single input stream and
 // copy the parameters from the stream to the context



  


  In my standalone minimal reproducer this always results on the codec ID of the single stream being AV_CODEC_ID_PCM_F32LE, in both macOS and Windows. When I integrate this code in my Qt application on Windows, I get the same result. However, on macOS, most of the times results in the codec id of the stream being AV_CODEC_ID_PCM_S16LE (via AV_CODEC_ID_FIRST_AUDIO) and sometimes AV_CODEC_ID_PCM_F32LE. Both sample formats are supported by my microphone.

  


  AV_CODEC_ID_PCM_F32LE always results in a correct output. AV_CODEC_ID_PCM_S16LE results on buzzy noisy audio slowed down to 0.5x, and If in this case I decode with AV_CODEC_ID_PCM_F32LE instead of copying the codec parameters from the stream, the output sounds correct again.

  


  I am trying to write generic code, so while enforcing the AV_CODEC_ID_PCM_F32LE codec works, I'd rather understand what is happening.

  


  What am I missing ? Is Qt interacting in some way that I can't think of ? I am compiling and linking my own ffmpeg libraries (6.1.1) and not using Qt's ones.

  


• Clip long video segment quickly

  30 janvier 2020, par PRMan

  Let’s say I have a video called Concert.mp4. I want to extract a performance from it quickly with minimal reencoding. I want to do the equivalent of this, but faster :

  ffmpeg -i "Concert.mp4" -ss 00:11:45 -to 00:18:15  -preset veryfast -y artist.mp4

  This takes 17 seconds, which is way too long for our needs.

  Now, it turns out that 11:45 and 18:15 don’t fall on iframes, so if you try this you will get a 3 second delay at the beginning before the video shows :

  ffmpeg -i "Concert.mp4" -ss 00:11:45 -to 00:18:15 -c copy -y artist.mp4

  Running this command, we can see where we need to cut :

  ffprobe -read_intervals "11:00%19:00" -v error -skip_frame nokey -show_entries frame=pkt_pts_time -select_streams v -of csv=p=0 "Concert.mp4" > frames.txt

  So what we need to do is encode the first 3.708 seconds, copy the middle, and then encode the last 5.912 seconds.

  I can get the 3 segments to all look perfect (by themselves) like this :

  ffmpeg -ss 698.698 -i "Concert.mp4" -ss 6.302 -t 3.708 -c:v libx264 -c:a copy -c:s copy -y clipbegin.mp4
  
  
  
  ffmpeg -ss 708.708 -to 1089.088 -i "Concert.mp4" -c copy -y clipmiddle.mp4
  
  
  
  ffmpeg -ss 1089.088 -i "Concert.mp4" -t 5.912 -c:v libx264 -c:a copy -c:s copy -y clipend.mp4
  
  
  
  ffmpeg -f concat -i segments.txt -c copy -y artist.mp4

  segments.txt of course contains the following :

  file 'clipbegin.mkv'
  
  file 'clipmiddle.mkv'
  
  file 'clipend.mkv'

  I saw this solution presented here, but no amount of tweaking gets it to work for me :

  https://superuser.com/a/1039134/73272

  As far as I can tell, this method doesn’t work at all. It crashes VLC pretty hard no matter what I try.

  The combined video keeps glitching after the 3 seconds, probably because the PTS times are different or something (using some options, I have seen warning messages to this effect). Is there anything I can add to the commands above to get this to work ? The only requirement is that the middle command must not re-encode the video, but must do a fast copy.

  Thanks in advance.

• FFmpeg chromakey libavfilter using C-API - key/replace green with alpha transparency

  3 juin 2020, par ZeroDefect

  I'm trying to use the FFmpeg chromakey libavfilter (via the C-API using C++17) to key some green pixels in a YUVA422p image and replace them with alpha transparency.

  



  Now, I setup/initialise the graph, connect the filters, and push through a frame ; however, the output frame appears unchanged. I suspect one of my configuration parameters is incorrect, but I'm really unsure. After having read the pertinent documentation, I still don't understand the problem.

  



  I have published a (minimal) code sample in github - https://github.com/zerodefect/chromakey_test. I have tried to keep the code sample as brief as possible but it is still a bit lengthy.

  



  The code sample includes a sample image (green_screen.png) for the purposes of testing.

  



  To run the application, the following parameters are required :

  



  


  ./cb_chroma_key_test ./green_screen.png [OUTPUT_PATH]

  


  



  The application dumps out a PLANAR image at YUV422p which I then load in via rawpixels.net - a brilliant little online utility to view raw image data (packed or planar).

  



  My avfilter graph consists of :

  



  


  buffersrc -> format -> chromakey -> buffersink

  


  



  The format filter is taking the RGBA (packed) format and converting it to YUVA422 planar.

  



  

  • GCC 8.4

  


  • Ubuntu 18.04

  


  • FFmpeg 4.2