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

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• JavaFX won't play mp3s with non integer bitrates

  While making a simple music player using JavaFX I realized that it wouldn't play certain mp3s I have. After looking closer into what the difference was between what played and what didn't I discovered that Java wouldn't play the songs with bitrates like 320.039kbs, but songs that were an even 320kbs would work just fine. Is there anything I can do about this?
• Why can't I use the STDdraw and STDAudio classes at the same time in java?

  I am working a program that plays sounds by clicking keyboard keys. I have gotten it to work, however when I try to visualize that sound, The sound stops playing. So I can either play the sounds, or make the waves but not both. How do I make them both work?

      public class GuitarHeroVisualizer { 
       public static void main(String[] args) {
        String keyboard = "q2we4r5ty7u8i9op-[=zxdcfvgbnjmk,.;/' ";
        GuitarString[] strings = new GuitarString[37];
        int x = 0;
        for (int i = 0; i < 37; i++) {
          strings[i] = new GuitarString(440.0 * Math.pow(1.05956, i - 23));
        }
        while (true) {
          if (StdDraw.hasNextKeyTyped()) {
              char key = StdDraw.nextKeyTyped();
              int j = keyboard.indexOf(key);
              if (j != -1) {
                  strings[j].pluck();
              }
          }
           // compute the superposition of samples
           //double sample = stringA.sample() + stringC.sample();
           double sample = 0;
           for (int i = 0; i < 37; i++) {
              sample += strings[i].sample();
           }
  
          // play the sample on standard audio
  
          if(sample != 0)
          {
              StdDraw.setXscale(0, 90);
              StdDraw.line(x, 0, x + 1, sample);
              StdDraw.line(x + 1, sample, x + 2, 0);
              x += 2;
          }
  
          StdAudio.play(sample);
  
          for (int i = 0; i < 37; i++) {
              strings[i].tic();
          }
      }
   }
  

  }

  Other two classes:

      public class GuitarString {
  RingBuffer buffer;
  int N;
  int time = 0;
  public GuitarString(double frequency) {
      N = (int)(44100/frequency);
      buffer = new RingBuffer(N);
      for (int i = 0; i < N; i++) {
          buffer.enqueue(0);
      }
  }
  
  public GuitarString(double[] init) {
      buffer = new RingBuffer(init.length + 1);
      N = init.length + 1;
      for (int i = 0; i < init.length; i++) {
          buffer.enqueue(init[i]);
      }
  }
  
  public void pluck() {
      for (int i = 0; i < N; i++) {
          buffer.dequeue();
      }
      for (int i = 0; i < N; i++) {
          buffer.enqueue(Math.random()-0.5);
      }
  }
  
  public void tic() {
      time++;
      double first = buffer.dequeue();
      double avg = (buffer.peek() + first) / 2;
      buffer.enqueue(avg * .994);
  }
  
  public double sample() {
      return buffer.peek();
  }
  
  public int time() {
      return time;
  }
  

  }

  next:

      public class RingBuffer {
  double[] nums;
  int capacity;
  int size = 0;
  int first = 0;
  int last = 0;
  public RingBuffer(int cap) {
      capacity = cap;
      nums = new double[cap];
  }
  
  public int size() {
      return size;
  }
  
  public boolean isEmpty() {
      if (size() == 0) {
          return true;
      }
      return false;
  }
  
  public boolean isFull() {
      if (size() == capacity) {
          return true;
      }
      return false;
  }
  
  public void enqueue(double x) {
      if (isFull()) {
          throw new RuntimeException("Queue is full.");
      }
      nums[last] = x;
      last++;
      if(last == capacity) {last = 0;}
      size++;
  }
  
  public double dequeue() {
      if (isEmpty()) {
          throw new RuntimeException("Queue is empty.");
      }
      double num = nums[first];
      first++;
      if(first == capacity) {first = 0;}
      size--;
      return num;
  }
  
  public double peek() {
      return nums[first];
  }
  

  }

  here is the std classes: https://introcs.cs.princeton.edu/java/stdlib/javadoc/StdDraw.html https://introcs.cs.princeton.edu/java/stdlib/javadoc/StdAudio.html
• How can I play FairPlay-encrypted audio on an AppleWatch with watchOS 5?

  watchOS 5 supports a new background audio mode so that audio apps (Audible, Overcast, etc) can play from the watch directly. I'm working on audio functionality for my app, but I have to use FairPlay DRM. I have segmented aac audio files with .m3u8 playlists, and I'm downloading them using AVAssetDownloadTask and do FairPlay key stuff with AVAssetResourceLoaderDelegate on the iPhone. And I'd like to add a watch app to play my content without the phone present.

  But I don't see anything like this on the watch APIs. AVAssetResourceLoaderDelegate and AVContentKeySession aren't available on watchOS, and AVAudioPlayer doesn't take an AVAsset or seemingly any other way to do FairPlay key stuff (I use AVPlayer on the phone).

  In the Creating Audio Apps for watchOS session at WWDC 2018, the presenter says:

  And in the case of AVAudioEngine, you can actually play some DRM content in conjunction with AVAudioPlayer node. And so you can now play your own DRM content, decrypt it yourself, and play it back for your user.

  But I haven't found anything yet about AVAudioEngine and FairPlay audio.

  I've looked into AVAudioExportSession to try to transcode my FairPlay content to MP3 or AAC or M4A or something that I can transfer to the watch, but it doesn't think any formats are compatible:

  AVAsset *asset = [self getDownloadedFairplayAsset];
  NSArray<NSString *> *possible = [AVAssetExportSession exportPresetsCompatibleWithAsset:asset];
  
      for (NSString *preset in possible)
      {
          AVAssetExportSession *session = [AVAssetExportSession exportSessionWithAsset:asset presetName:preset];
          [session determineCompatibleFileTypesWithCompletionHandler:^(NSArray<AVFileType> * _Nonnull compatibleFileTypes) {
              NSLog(@"Compatible types with %@: %@", preset, compatibleFileTypes);
              //this is always an empty array
          }];
      }
  

  So, my question: How can I play FairPlay-encrypted audio on a watchOS 5 app?
• Route ASIO input to WDM virtual microphones

  I'm doing some sound manipulation with the Dante Virtual Soundcard (DVS). The software I am using (ffmpeg) can only record from WDM microphones. DVS has a limitation of 8 stereo channels for WDM, so I'm seeing if I can use the 64 channel ASIO option.

  Basically, I am trying to route 64 ASIO input channels to 32 virtual stereo WDM microphones. I would then initiate ffmpeg recordings of these 32 WDM microphones as different times depending on which microphones are used.

  NAudio's patchbay example (https://github.com/markheath/naudio-asio-patchbay) was a good start but routes to ASIO outputs. Is it possible to route to multiple WaveIn devices?
• Alternative for NAudio, an audio library, in .net core 2.1 for Linux

  We were wondering if there is an audio library for .net core 2.1 similar to the library NAudio but compatible also in Linux.

  Thank you so much in advance.
• NAudio playing music before function is even called

  WaveOutEvent player = new WaveOutEvent();
  WaveChannel32 volumeChannel = new WaveChannel32(waveStream);
  player.Init(volumeChannel);
  
  Multimedia.Timer musicdelayer = new Multimedia.Timer();    
  
  musicdelayer.Mode = Multimedia.TimerMode.OneShot;
  musicdelayer.Tick += Musicdelayer_Tick;
  musicdelayer.Period = (offset < 0) ? Math.Abs(offset) : 1;
  

  Tick Event:

  private void Musicdelayer_Tick(object sender, EventArgs e)
  {
          Console.WriteLine("tick");
          player.Play();
  }
  

  Now something incredibly strange happens:

  The console outputs "tick" 5 seconds after the tick event is called with the offset of -5000... but the player has started playing already instantly before tick is even called.. what the hell?

  I am just confused out of my mind to how this is happening. Any idea what might be going on?
• How can I subscribe on c# stream from python using pythonnet?

  I need to analyze loopbackcapture stream with python.

  I didn't found any python wrappings for wasapi loopbackcapture, so I had to use c# (in which I don't have any experience).

  For now I have c# assembly dll which is wrappings to wasapiLoopbackCapture and records this to the file. But I need to perform analysis on the real time.

   using (WasapiCapture soundIn = new WasapiLoopbackCapture())
          {
  
              //initialize the soundIn instance
              soundIn.Initialize();
  
              //create a SoundSource around the the soundIn instance
              SoundInSource soundInSource = new SoundInSource(soundIn) { FillWithZeros = false };
  
              //create a source, that converts the data provided by the soundInSource to any other format
              IWaveSource convertedSource = soundInSource
                  .ChangeSampleRate(sampleRate) // sample rate
                  .ToSampleSource()
                  .ToWaveSource(bitsPerSample); //bits per sample
  
              //channels...
              convertedSource = convertedSource.ToMono()
  
              //create a new wavefile
              WaveWriter waveWriter = new WaveWriter(output_file, convertedSource.WaveFormat)
              //register an event handler for the DataAvailable event of the soundInSource
              soundInSource.DataAvailable += (s, e) =>
                  {
                      //read data from the converedSource
                      byte[] buffer = new byte[convertedSource.WaveFormat.BytesPerSecond / 2];
                      int read;
  
                      //keep reading as long as we still get some data
                      while ((read = convertedSource.Read(buffer, 0, buffer.Length)) > 0)
                      {
                          //write the read data to a file
                          waveWriter.Write(buffer, 0, read);
                      }
                  };
  }
  

  1) Is it possible to return stream from c# and subscribe on it from python?

  2) another variant is to analyze stream on c# side and fire some events to python script. how can I do this? ( I mean event's firing and handling side of thing)
• WASAPI capture data is only 00 and FF

  If I open the captured PCM data with a Hex eitor they are only combinations of 00 and FF bytes.It seems to capture only static noise on my laptop and pc.On my pc, the recording device (Stereo Mix - Realtek High Definiton Audio) and on my laptop I use the recording device Stereo Mix (Realtek Audio) .I can modify the WAVEFORMATEX and change the driver format but I still only get different kind of loud static noises or no sound at all...If I use Audacity to capture, it works, so the problem must be at my code...

       HRESULT InitAudio() {
  HRESULT hr = E_FAIL;
  REFERENCE_TIME hnsRequestedDuration;
  REFERENCE_TIME hnsActualDuration;
  UINT32 bufferFrameCount;
  IMMDeviceEnumerator *pEnumerator = NULL;
  IMMDeviceCollection* pIMMDeviceCollection = NULL;
  IMMDevice *pDevice = NULL;
  _AudioSamplesReadyEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE);
  _ShutdownEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE);
  
      WAVEFORMATEX                wfx =
    {WAVE_FORMAT_PCM, wavFile.num_channels, wavFile.sample_rate, wavFile.byte_rate, wavFile.sample_alignment, wavFile.bit_depth, 0};
  
  hr = CoCreateInstance(
         CLSID_MMDeviceEnumerator, NULL,
         CLSCTX_ALL, IID_IMMDeviceEnumerator,
         (void**)&pEnumerator);
  
  hr = pEnumerator->EnumAudioEndpoints(eCapture,DEVICE_STATE_ACTIVE,&pIMMDeviceCollection);
  hr = pIMMDeviceCollection->Item(0, &pDevice);
  
  hr = pDevice->Activate(
                  IID_IAudioClient, CLSCTX_ALL,
                  NULL, (void**)&pAudioClient);
  
  hr = pAudioClient->GetDevicePeriod(NULL, &hnsRequestedDuration);
  hr = pAudioClient->Initialize(
                       AUDCLNT_SHAREMODE_SHARED,
                       AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
                       hnsRequestedDuration,
                       0,
                       &wfx,
                       NULL);
  
  
  hr = pAudioClient->SetEventHandle(_AudioSamplesReadyEvent);
  if (FAILED(hr)) MessageBox(NULL,TEXT("Unable to set event handle!"), TEXT("Error"), MB_OK);
  
  hr = pAudioClient->GetService(
                       IID_IAudioCaptureClient,
                       (void**)&pCaptureClient);
  
  ProcessSoundThread = CreateThread(NULL, 0, ProcessSound, 0, 0, NULL);
  
  hr = pAudioClient->Start();  // Start recording.
  
  return hr;
  }
  

  and the function that is used to process the data

      DWORD WINAPI ProcessSound(void* arg) {
  UINT32 numFramesAvailable;
  UINT32 packetLength = 0;
  UINT32 dataSize = 0;
  BYTE *pData;
  DWORD  flags;
  HRESULT hr;
  UINT dwDataWrote;
  
  HANDLE waitArray[2] = { _AudioSamplesReadyEvent, _ShutdownEvent};
  HANDLE mmcssHandle = NULL;
  DWORD mmcssTaskIndex = 0;
  
  hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
  if (FAILED(hr)) MessageBox(NULL,TEXT("Unable to initialize COM in render thread!"), TEXT("Error"), MB_OK);
  
  mmcssHandle = AvSetMmThreadCharacteristics("Audio", &mmcssTaskIndex);
  if(mmcssHandle == NULL) MessageBox(NULL,TEXT("Unable to enable MMCSS on capture thread!"), TEXT("Error"), MB_OK);
  
  while(captureSound) {
      DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
      switch (waitResult)
          {
          case WAIT_OBJECT_0 + 0:
              hr = pCaptureClient->GetBuffer(&pData,&numFramesAvailable,&flags,NULL,NULL);
              if(hr != AUDCLNT_S_BUFFER_EMPTY) {
                  dataSize = numFramesAvailable * wavFile.sample_alignment; // wavFile.sample_alignment = frameSize = num_channels * (pWav->bit_depth)/8
                  if (FAILED( addWavData(&wavFile, pData, dataSize, &dwDataWrote)))
                      MessageBox(NULL,TEXT("Error writting the data!"), TEXT("Error"), MB_OK); 
                  totalData += dataSize;
              }
              pCaptureClient->ReleaseBuffer(numFramesAvailable);
              break;
          case WAIT_OBJECT_0 + 1:   
              captureSound = false;
              break;
          }`
  }
  return 0;
  }
  
• WASAPI captured data is only 00 and FF

  If I open the captured PCM data they are only combinations of 00 and FF bytes.It seems to capture only static noise on my laptop and pc.On my pc, the recording device (Stereo Mix - Realtek High Definiton Audio) and on my laptop I use the recording device Stereo Mix (Realtek Audio) .I can modify the WAVEFORMATEX and change the driver format but I still only get different kind of loud static noises or no sound at all...If I use Audacity to capture, it works, so the problem must be at my code.

      HRESULT InitAudio() {
  HRESULT hr = E_FAIL;
  REFERENCE_TIME hnsRequestedDuration;
  REFERENCE_TIME hnsActualDuration;
  UINT32 bufferFrameCount;
  IMMDeviceEnumerator *pEnumerator = NULL;
  IMMDeviceCollection* pIMMDeviceCollection = NULL;
  IMMDevice *pDevice = NULL;
  _AudioSamplesReadyEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE);
  _ShutdownEvent = CreateEventEx(NULL, NULL, 0, EVENT_MODIFY_STATE | SYNCHRONIZE);
  
      WAVEFORMATEX                wfx =
    {WAVE_FORMAT_PCM, wavFile.num_channels, wavFile.sample_rate, wavFile.byte_rate, wavFile.sample_alignment, wavFile.bit_depth, 0};
  
  hr = CoCreateInstance(
         CLSID_MMDeviceEnumerator, NULL,
         CLSCTX_ALL, IID_IMMDeviceEnumerator,
         (void**)&pEnumerator);
  
  hr = pEnumerator->EnumAudioEndpoints(eCapture,DEVICE_STATE_ACTIVE,&pIMMDeviceCollection);
  hr = pIMMDeviceCollection->Item(0, &pDevice);
  
  hr = pDevice->Activate(
                  IID_IAudioClient, CLSCTX_ALL,
                  NULL, (void**)&pAudioClient);
  
  hr = pAudioClient->GetDevicePeriod(NULL, &hnsRequestedDuration);
  hr = pAudioClient->Initialize(
                       AUDCLNT_SHAREMODE_SHARED,
                       AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST,
                       hnsRequestedDuration,
                       0,
                       &wfx,
                       NULL);
  
  
  hr = pAudioClient->SetEventHandle(_AudioSamplesReadyEvent);
  if (FAILED(hr)) MessageBox(NULL,TEXT("Unable to set event handle!"), TEXT("Error"), MB_OK);
  
  hr = pAudioClient->GetService(
                       IID_IAudioCaptureClient,
                       (void**)&pCaptureClient);
  
  ProcessSoundThread = CreateThread(NULL, 0, ProcessSound, 0, 0, NULL);
  
  hr = pAudioClient->Start();  // Start recording.
  
  return hr;
  }
  

  and the function that is used to process the data

      DWORD WINAPI ProcessSound(void* arg) {
  UINT32 numFramesAvailable;
  UINT32 packetLength = 0;
  UINT32 dataSize = 0;
  BYTE *pData;
  DWORD  flags;
  HRESULT hr;
  UINT dwDataWrote;
  
  HANDLE waitArray[2] = { _AudioSamplesReadyEvent, _ShutdownEvent};
  HANDLE mmcssHandle = NULL;
  DWORD mmcssTaskIndex = 0;
  
  hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
  if (FAILED(hr)) MessageBox(NULL,TEXT("Unable to initialize COM in render thread!"), TEXT("Error"), MB_OK);
  
  mmcssHandle = AvSetMmThreadCharacteristics("Audio", &mmcssTaskIndex);
  if(mmcssHandle == NULL) MessageBox(NULL,TEXT("Unable to enable MMCSS on capture thread!"), TEXT("Error"), MB_OK);
  
  while(captureSound) {
      DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
      switch (waitResult)
          {
          case WAIT_OBJECT_0 + 0:
              hr = pCaptureClient->GetBuffer(&pData,&numFramesAvailable,&flags,NULL,NULL);
              if(hr != AUDCLNT_S_BUFFER_EMPTY) {
                  dataSize = numFramesAvailable * wavFile.sample_alignment; // wavFile.sample_alignment = frameSize = num_channels * (pWav->bit_depth)/8
                  if (FAILED( addWavData(&wavFile, pData, dataSize, &dwDataWrote)))
                      MessageBox(NULL,TEXT("Error writting the data!"), TEXT("Error"), MB_OK); 
                  totalData += dataSize;
              }
              pCaptureClient->ReleaseBuffer(numFramesAvailable);
              break;
          case WAIT_OBJECT_0 + 1:   
              captureSound = false;
              break;
          }`
  }
  return 0;
  }