NAudio | Audio and MIDI library for NET | Audio Utils library

I'm trying to install a package that's on nuget.

This one: https://www.nuget.org/packages/NAudio/

When I follow this guide: https://docs.microsoft.com/en-us/nuget/quickstart/install-and-use-a-package-in-visual-studio about how you add packages to your project I'm stuck at 2) since I only have "Microsoft Visual Studio Offline Packages" available as a source.

This is a new WPF project on a fresh install of VS.net 2019 community edition.

Any ideas what I'm doing wrong?

I am trying to convert .vox to .mp3 or .wav with NAudio with the code below:-

Is there any way to play audio directly into a capture device in C#? In my project I will have to feed later on a virtual capture driver with audio so I can use it in other programs and play the wanted audio anywhere else, but Im not sure it is possible in C#, I tried to do this with NAudio (which is truly amazing):

I am using websocket to call received, accept and all. I am getting websocket event when call accepted.

websocket accept event output :-

I want to change windows 10 default audio output with NAudio.

NAudio has an api to get the default audio endpoint:

I would like to know how can I split the channels of a WAV file into two byte arrays with the PCM data.

I've been trying to do this with NAudio, but I can't get it.

So I've written a program in C# that gets the current available audio output devices. So when i run the process I get the names of the devices inside the DataReceived event. When it receives "DONE" it kills the process and adds the saved names to the TMP_Dropdown options. but the problem is that when it gets to dropdown.ClearOptions() the program just stops without any error messages. When i add a breakpoint and just keep stepping through that function the yellow bar just dissapears and the function just stops. But when i just add some random strings to devices and don't run GetDevices() it works like a charm.

here is the code I was referencing above:

I'm having an issue with a BufferedWaveProvider from NAudio library. I'm recording 2 audio devices (a microphone and a speaker), merge them into 1 stream and send it to an encoder (for a video).

To do this, I do the following:

1. Create a thread where I'll record the microphone using WasapiCapture.
2. Create a thread where I'll record the speakers audio using WasapiLookbackCapture. (I also use a SilenceProvider so I don't have gaps in what I record).
3. I'll want to mix these 2 audio so I have to make sure they have the same format, so I detect what's the best WaveFormat in all these audio devices. In my scenario, it's the speaker. So I decide that the Microphone audio will pass through a MediaFoundationResampler to adapt its format so it has the same than the one from the speaker.
4. Each audio chunks from the Wasapi(Lookback)Capture are sent to a BufferedWaveProvider.
5. Then, I also made a MixingSampleProvider where I pass the ISampleProvider from each recording thread. So I'm passing the MediaFoundationResampler for the Microphone, and BufferedWaveProvider for the Speakers.
6. In loop in a third thread, I read the data from the MixingSampleProvider, which is supposed to asynchronously empty the BufferedWaveProvider(s) while it's getting filled.
7. Because each buffer may not get filled exactly at the same time, I'm looking at what's the minimal common duration there is between these 2 buffers, and I'm reading this amount out of the mixing sample provider.
8. Then I enqueue what I read so my encoder, in a 4th thread, will treat it in parallel too.

Please see the flowchat below that illustrates my description above.

• It works GREAT when recording the microphone and speaker for more than 1h while playing video game that uses the microphone too (for online multiplayer). No crash. The buffers are staying quite empty all the time. It's awesome.
• But for some reason, every time I try my app during a Discord, Skype or Teams audio conversation, I immediately (within 5sec) crash on BufferedWaveProvider.AppSamples because the buffer gets full.

Looking at it in debug mode, I can see that:

• The buffer corresponding to the speaker is almost empty. It has in average 100ms max of audio.
• The buffer corresponding to the microphone (the one I resample) is full (5 seconds).

From what I read on NAudio's author's blog, the documentation and on StackOverflow, I think I'm doing the best practice (but I can be wrong), which is writing in the buffer from a thread, and reading it in parallel from another one. There is of course a risk that it's getting filled faster than I read it, and it's basically what's happening right now. But I'm not understanding why.

I'd like some help to understand what I'm missing here, please. The following points are confusing me:

1. Why does this issue happens only with Discord/Skype/Teams meetings? The video games I'm using are using the microphone too, so I can't imagine it's something like another app is preventing the microphone/speakers to works correctly.

2. I synchronize the startup of both audio recorder. Do to this, I'm using a signal to ask the recorders to starts, and when they all started to generate data (through DataAvailable event), I send a signal to tell them to fill in the buffers with what they will receive in the next event. It's probably not perfect because both audio devices send their DataAvailable at different times, but we're talking about 60ms of difference maximum (on my machine), not 5 seconds. So I don't understand why it's getting filled.

3. To bounce on what I said in #2, my telemetry shows that the buffer is getting filled this way (values are dummy):

I'm creating a chat client that has UDP for voice chat, but when I send the audio (in bytes) to the other clients, the client plays the audio everything is clear but I hear a random clicking sound in the back ground. I thought it might be because it's UDP and not checking if the data is correct, but no. Even when I send through TCP I can still hear the clicking sound in the background.

CODE to recreate: