No Audio In When Not Default Device And Sound Control Panel Closed

See also questions close to this topic

• ffmpeg - output file wrong

  I'm using ffmpeg to capturing the audio from an usb tv tunner on ubuntu.

  The command is:

  ffmpeg -i /dev/dvb/adapter0/dvr0 -ac 1 -ar 11025 -f segment -segment_time 15 -vn -reset_timestamps 1 -strftime 1 "out/ref_001_dvr0_%Y-%m-%d_%H-%M-%S.wav"
  

  The problem is in the output files (image attahed). output files As you can see in the image we have files with extension wav and mp3 but in the ffmpeg sentence just should to be created wav files.

  I dont have idea whats is wrong in my ffmpeg sentence. Any ideas?
• FFmpeg split filter with with audio filtering throws errors

  When using FFmpeg's split filter for video tracks, I want to filter audio track as well. I tested asplit but not sure where to use it in the filter chain.

  When running this command:
  ffmpeg -y -probesize 100M -analyzeduration 5000000 -hide_banner -i $input -i $logo \ -filter_complex "[0:a]aformat=channel_layouts=stereo,aresample=async=1000[a1];[0:v]overlay=20:20,drawtext=fontfile=$font:text='some text':fontcolor=c1ff30:fontsize=50:x=250:y=100,split=3[v1][v2][v3];[v1]setpts=PTS-STARTPTS,yadif=0:-1:0,scale=w=640:h=360:force_original_aspect_ratio=decrease:sws_dither=ed:flags=lanczos,setdar=16/9[v1];[v2]setpts=PTS-STARTPTS,yadif=0:-1:0,scale=w=1024:h=576:force_original_aspect_ratio=decrease:sws_dither=ed:flags=lanczos,setdar=16/9[v2];[v3]setpts=PTS-STARTPTS,yadif=0:-1:0,scale=w=1600:h=900:force_original_aspect_ratio=decrease:sws_dither=ed:flags=lanczos,setdar=16/9[v3]" \ -map "[v1]" -map "[a1]" -c:a libfdk_aac -ac 2 -b:a 128k -ar 48000 -c:v libx264 -crf 23 -maxrate 550k -bufsize 1100k -bsf:v h264_mp4toannexb -forced-idr 1 -sc_threshold 0 -r 25 -g 50 -keyint_min 50 -preset medium -profile:v main -level 3.1 -coder 1 -pix_fmt yuv420p -flags +loop+mv4+cgop -flags2 +local_header -movflags faststart -cmp chroma -hls_time 6 -hls_playlist_type vod /dir/1.m3u8 \ -map "[v2]" -map "[a1]" -c:a libfdk_aac -ac 2 -b:a 128k -ar 48000 -c:v libx264 -crf 23 -maxrate 1400k -bufsize 2800k -bsf:v h264_mp4toannexb -forced-idr 1 -sc_threshold 0 -r 25 -g 50 -preset medium -profile:v main -level 4 -coder 1 -pix_fmt yuv420p -flags +loop+mv4+cgop -flags2 +local_header -movflags faststart -cmp chroma -keyint_min 50 -hls_time 6 -hls_playlist_type vod /dir/2.m3u8 \ -map "[v3]" -map "[a1]" -c:a libfdk_aac -ac 2 -b:a 128k -ar 48000 -c:v libx264 -crf 23 -maxrate 3100k -bufsize 6200k -bsf:v h264_mp4toannexb -forced-idr 1 -sc_threshold 0 -r 25 -g 50 -preset medium -profile:v high -level 3.1 -coder 1 -pix_fmt yuv420p -flags +loop+mv4+cgop -flags2 +local_header -movflags faststart -cmp chroma -keyint_min 50 -hls_time 6 -hls_playlist_type vod /dir/3.m3u8

  FFmpeg throws this error:
  

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/aaa/bbb/file.mov':
    Metadata:
      major_brand     : qt  
      minor_version   : 512
      compatible_brands: qt  
      encoder         : Lavf58.20.100
    Duration: 00:00:10.00, start: 0.000000, bitrate: 117945 kb/s
      Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le(tv, bt709, top coded first (swapped)), 1920x1080, 115636 kb/s, SAR 1:1 DAR 16:9, 25 fps, 25 tbr, 10k tbn, 10k tbc (default)
      Metadata:
        handler_name    : Telestream, LLC Telestream Media Framework - Local 99.99.999999
        encoder         : Apple ProRes 422
        timecode        : 01:25:44:05
      Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, stereo, s32 (24 bit), 2304 kb/s (default)
      Metadata:
        handler_name    : Telestream, LLC Telestream Media Framework - Local 99.99.999999
      Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
      Metadata:
        handler_name    : Telestream, LLC Telestream Media Framework - Local 99.99.999999
        timecode        : 01:25:44:05
  Input #1, png_pipe, from '/Volumes/aaa/bbb/logo.png':
    Duration: N/A, bitrate: N/A
      Stream #1:0: Video: png, rgba(pc), 1920x1080 [SAR 2835:2835 DAR 16:9], 25 tbr, 25 tbn, 25 tbc
  Output with label 'a1' does not exist in any defined filter graph, or was already used elsewhere.
  

  When removing the audio filtering ([0:a]aformat=channel_layouts=stereo,aresample=async=1000[a1]) and mapping 0:a as audio, the command runs fine.

  What am I missing?
• Audio isn't playing in js

  I've been having this problem where audio wouldn't play. I am trying to get a random sound to play but nothing happens when I click on the image (which has an onclick="playRandomSound()"). This is my code.

  var sound1 = new Audio("InDeed.wav");
  var sound2 = new Audio("Indeed2.wav");
  var sound3 = new Audio("Yeeees.wav");
  var sound4 = new Audio("girl_shut_up.wav");
  var sound5 = new Audio("imhmmm.wav");
  var randomizer;
  function playRandomSound(){
  randomizer = Math.floor(Math.random()*5);
  if (randomizer === 0){
    sound1.play();  
  } else if (randomizer == 1){
    sound2.play();
  } else if (randomizer == 2){
    sound3.play();
  } else if (randomizer == 3){
    sound4.play();
  } else if (randomizer == 4){
    sound5.play();
  }
  }

  body {
    background: white;
  }
  #rokas {
    height: 100vh;
  }

  <HTML>
    <HEAD>
      
    </HEAD>
    <BODY>
      <img src="https://i.imgur.com/4lO3dpT.jpg" id="rokas" onclick="playRandomSound()">
    </BODY>
  </HTML>

  enter code here
• I want to make instant sound decibel meter in c#

  want to make sound meter without using arduino and want to show this value in textbox... if you help me to get a value of instant decibels, I will do the rest..I started the first time .. I found a NAudio sound library but I can't intervene..Thanks for your help :))
• How to get an array from an audio file and then get the audio file back?

  I have an audio file with a duration of 1 second in the format".wave".I need to get an array out of it. Then I'm going to change the data in the array, but I don't know how to get this array and then get the audio file from the modified array. Read about NAUDIO.dll. Tried to do with the following code from the Internet:

  using (WaveFileReader reader = new WaveFileReader(path))
  {
      Assert.AreEqual(16, reader.WaveFormat.BitsPerSample, "Only works with 16 bit audio");
      byte[] buffer = new byte[reader.Length];
      int read = reader.Read(buffer, 0, buffer.Length);
      short[] sampleBuffer = new short[read / 2];
      Buffer.BlockCopy(buffer, 0, sampleBuffer, 0, read);
  }
  

  but produces the following error:

  Additional information: Not a WAVE file - no RIFF heade

  I don't know what to do.can someone help me here?
• C# NAudio FFT on WASAPI audio capture

  This is what I have so far:

  private void capture()
      {
          var capture = new WasapiLoopbackCapture();
          capture.DataAvailable += (s, a) =>
          {
              if (stop)
              {
                  capture.StopRecording();
              }
          };
          capture.RecordingStopped += (s, a) =>
          {
              capture.Dispose();
          };
          capture.StartRecording();
          while (capture.CaptureState != NAudio.CoreAudioApi.CaptureState.Stopped)
          {
              Thread.Sleep(500);
          }
      }
  

  I'd like to do FFT on a WASAPI audio capture, and then get the amplitude of a certain frequency band.
• WASAPI - Exact time of render

  Is there a way to know the exact timestamp of when an audio packet has been rendered by the endpoint?

  If I have something like this:

  hr = pSelf->mpAudioRenderclient->GetBuffer(nFrames, &_dest);
  ASSERT(hr == S_OK);
  if (S_OK == hr)
  {
      while (_length > 0)
      {
          //... some manipulation on _dest
      }
      hr = pSelf->mpAudioRenderclient->ReleaseBuffer(nFrames, 0);
      gettimeofday(&now, NULL);
      ASSERT(hr == S_OK);
  }
  

  Is it as soon as ReleaseBuffer() has returned, or are there a few ms that the system might take before it renders this specific audio packet?
• How to capture audio on c++ UWP app with WASAPI?

  I can not seem to find any tutorial on the internet for my question.
  All the simple guide is not suitable for UWP.

  For Example, To use WASAPI there are these steps

  • enumerate devices
  • capture audio
  • play (render) audio back

  But the enumerating step, The client must call CoCreateInstance. But from my understanding this function is not support in UWP. Also I failed at Line 30 when following this code.

  So, I try to understand This, C++ UWP using WASAPI, But I can't find any Enumerate part and this project is very complicate for me. It include a lot of other files (DeviceState.h, common.h)
  And I failed to extract the code to create my own application.

  My question is how can I capture audio on c++ UWP app with WASAPI?

  If this question is too board, I will change my question to How to enumerate audio device in c++ UWP application?.

  And the reason why I use WASAPI is because I want to access the data stored in the Buffer.
• Continuous WASAPI Ring-Buffer Sampling

  How to use WASAPI (or something like it) to continuously sample audio into a (thread-safe) ring-buffer, so that a consumer thread can read from that buffer in an a set interval?

  Currently we have a .sample() method that returns a chunk of samples after a set sampling interval, but this has quite the overhead due to memory allocation etc.. maybe this method could be optimized; I'm pretty sure we're doing it wrong.

  std::vector<short> sampler2::sample()
  {
      // prepare header
      waveInPrepareHeader(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));
  
      // insert a wave input buffer
      waveInAddBuffer(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));
  
      // commence sampling input
      waveInStart(hWaveIn);
  
      // sleep for the duration of a sample interval
      std::this_thread::sleep_for(milliseconds(SAMPLE_INTERVAL));
  
      // create vector
      std::vector<short> samplesChunk(&waveIn[0], &waveIn[0] + NUMPTS);
  
      // return vector
      return samplesChunk;
  }
  

  GitHub Links: sampler2.h & sampler2.cpp

  The code is very shitty and we have no clue how to properly use WASAPI. Our goal was to (quickly) create a sampler class that can leverage a sampling interval of >10 ms.