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• avformat_open_input for 24-bit audio fails intermittently with avfoundation due to "audio format is not supported"

  27 septembre 2019, par NaderNader

  My application uses the ffmpeg APIs (avformat, avdevice, etc) to open a selected audio input for encoding. For audio inputs configured for 24-bit I can reliably open them the first time, but when I close and reopen that input later, the avformat_open_input() call fails due to "audio format is not supported". My testing shows that it never fails the first time after starting my, and has only about a 50% chance of success when reopening.

  The failure only occurs when I configure my "Built-in Microphone" audio input for 24-bit integer. 16-bit integer and 32-bit float work reliably. Changing the number of channels and sample rate has no effect.

  I have read the documentation and see that the proper way to free the resources after opening is to call avformat_close_input. The only way I have found to guarantee success is to only open the input once.

  I have written a test program to recreate these failures.

  int main() {
  
  
  
      avdevice_register_all();
  
  
  
      cout << "Running open audio test" << endl;
  
  
  
  
  
      int i;
  
      for(i = 0; i< 10; i++) {
  
  
  
          AVDictionary* options = NULL;
  
          AVInputFormat* inputFormat = av_find_input_format("avfoundation");
  
          if (!inputFormat) {
  
              cout << "avfoundation inputFormat=null" << endl;
  
          }
  
  
  
          AVFormatContext* formatContext = avformat_alloc_context();
  
          int result = avformat_open_input(&formatContext, ":1", inputFormat, &options);
  
          if (result < 0) {
  
              char error[256];
  
              av_strerror(result, error, sizeof(error));
  
              cout << "result=" << result << " " << error << endl;
  
          } else {
  
              cout << "input opened successfully" << endl;
  
          }
  
  
  
          sleep(1);
  
  
  
          avformat_close_input(&formatContext);
  
  
  
          sleep(1);
  
  
  
      }
  
  
  
      return 0;
  
  }

  I would expect the main loop to succeed each time but a typical output shows a high failure rate :

  Running open audio test

  input opened successfully
  
  [avfoundation @ 0x7fdeb281de00] audio format is not supported
  
  result=-5 Input/output error
  
  [avfoundation @ 0x7fdeb2001400] audio format is not supported
  
  result=-5 Input/output error
  
  [avfoundation @ 0x7fdeb2001400] audio format is not supported
  
  result=-5 Input/output error
  
  input opened successfully
  
  input opened successfully
  
  input opened successfully
  
  [avfoundation @ 0x7fdeb2068800] audio format is not supported
  
  result=-5 Input/output error
  
  input opened successfully
  
  input opened successfully

  I have tried increasing the sleep time between close and open to 5 seconds, but saw no difference in behavior. While this program runs, I can use the MacOS Audio MIDI Setup to change the microphone input configuration and watch the output change from all success to intermittent errors.

  The source of the failure is https://github.com/FFmpeg/FFmpeg/blob/master/libavdevice/avfoundation.m#L672

  It appears avfoundation.m internally is opening an input stream and grabbing an audio frame to determine the format, but the value returned is not valid sometimes, when the process has previously opened and closed that input.

  Am I not closing the resources properly ? Do I have a hardware issue specific to my macbook ?

  Additional Details :

  Tested MacBook Pro with MacOS Mojave 10.14.6
  Tested with Ffmpeg 3.4.1, 4.0, and 4.1

  list_devices :

  [AVFoundation input device @ 0x7f80fff066c0] AVFoundation video devices:
  
  [AVFoundation input device @ 0x7f80fff066c0] [0] FaceTime HD Camera
  
  [AVFoundation input device @ 0x7f80fff066c0] [1] Capture screen 0
  
  [AVFoundation input device @ 0x7f80fff066c0] AVFoundation audio devices:
  
  [AVFoundation input device @ 0x7f80fff066c0] [0] Behringer Duplex
  
  [AVFoundation input device @ 0x7f80fff066c0] [1] Built-in Microphone
  
  [AVFoundation input device @ 0x7f80fff066c0] [2] USB Audio CODEC

• ffmpeg how to do the "earrape" sound effect

  20 avril 2020, par Shalin Shah

  I was wondering how to do the earrape effect using ffmpeg where the audio just sounds completely destroyed. Here's an example :
https://www.youtube.com/watch?v=KiCmvQiAC8Q

  



  I've tried a bunch of combinations of different commands on ffmpeg and the closest I've gotten is the following (where I use the superequalizer and then make the volume super high) :

  



  import ffmpeg
(
    ffmpeg
    .input('shark.wav')
    .filter("superequalizer", 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20)
    .filter("volume", 10)
    .output('output_earrape.wav')
    .run()
)


  



  I'm using a python wrapper but here's the command line equivalent :

  



  ffmpeg -i shark.wav -af "superequalizer=1b=20:2b=20:3b=20:4b=20:5b=20:6b=20:7b=20:8b=20:9b=20:10b=20:11b=20:12b=20:13b=20:14b=20:15b=20:16b=20:17b=20:18b=20,volume=10" output_earrape.wav


  



  The problem with the above is that it doesn't do anything for files that aren't already super loud (such as recorded audio) and most of the time the audio actually just ends up clipping and then being super soft.

  



  Does anyone have suggestions on how to do this effect ? Thanks !

  


• Streaming video over named PIPE with limited "channel" bandwidth

  13 mars 2019, par fmagno

  I have a video container vid.mp4 that I want to play with ffplay through a named PIPE and be able to tweak the maximum bandwidth allowed by the "channel". Follows what I did :

  1. Create a named PIPE :

  mkfifo pipe_in

  2. Send the container to the pipe with a limited bandwidth (150kB/s) with the help of pipe viewer pv :

  cat vid.mp4 | pv -L 150k > pipe_in

  3. Play the video with ffplay :

  ffplay cache:./pipe_in

  My expectation : To watch the video come through immediately but slowly given the bandwidth constraint.

  What really happens : The video begins to show at normal speed only when command 2. finishes running.

  Thank you in advance !