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Of course, there were always ways of dealing with audio in previous versions of Windows. This includes WASAPI (Windows Audio Session API), a brand new API for capturing and rendering audio. The DAC can do its own high quality upsampling filtering, which is of course needed for normal operation.Windows Vista introduced a new series of APIs for audio, called CoreAudio. The whole point is to transport the audio bit perfect and then forward it to the DAC. I know there are people who tell you that you cannot hear this and that, but i think upsampling is not one of those problem-free operations. There are a few good algorithms tho if run with certain parameters.
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I certainly can hear differences between upsampling libraries and upsampling parameters. If it is using windows default resampling routines, it is not going to be good. Upsampling everything to 96khz just creates more problems and most likely decreases quality. Maybe using kernel streaming capture (instead of wasapi exclusive) would tell the current samplerate? Maybe somehow capturing straight from the output device (not the recording device) would help? One possibility is to run the Asio Bridge as a child process and grab the number from its window once every second or so. It is almost like it is purposely hidden from virtual audio applications. It is amazing how hard it is to get to know that one number. If i had that one number (samplerate) or any other number that be used to derive the current sample rate, then the problem would be solved. Can i tap into the source of that information? How is this information obtained programmagically using for instance c++? Or maybe there is a registry entry that can be read? Maybe there is a DLL i could link to and API to get the information? The Asio Bridge application seems to always know the correct samplerate. How is following of correct playback format in the capture side done with VB (hifi) Cable? I don't want resampling. I am pretty sure many would like to support this feature if it was just available.
![vt3 wasapi sample rate vt3 wasapi sample rate](https://www.listeninc.com/wp/media/SC19_WASAPI_AC621-705x505.png)
This is not a feature tho, its a problem. This is not done in the current audio middleware software (an app between VB-cable and audio hardware). When using VB (hifi) Cable the audio capturing needs to somehow follow the samplerate of the playback software. If i start a "masters" record in Tidal i need to randomly change the samplerate until the audio stops being buggy. Tidal does not even tell the current sample rate it outputs using Wasapi Exclusive and there are 88200hz and 96000hz mastered content in addition to 44100hz content. This means that the samplerate is not always the default samplerate defined in the audio device properties. Modern online audio services and players, such as Tidal, foobar2000, jRiver, support low level audio APIs.
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