How to stream audio from microphone to Google Speech-to-Text in Python using WebRTC

Question

I'm looking for documentation about transcribing audio streaming data coming from WebRTC using Google Cloud Speach-To-Text. I'm using aiortc as a library in Python to handle the video and audio stream coming from a client web app.

Here is a snippet of the class that I'm using to process the audio data.

class AudioTransformTrack(MediaStreamTrack):
        kind = "audio"
    
        def __init__(self, track):
            super().__init__()
            self.track = track
    
        async def recv(self):
            frame = await self.track.recv()
            data_np = frame.to_ndarray().astype(dtype='float32').reshape(1920, )
            # print("data_np.shape:", data_np.shape)
            y_16k = librosa.resample(data_np, 48000, 16000)
            audio_data = y_16k.astype(dtype='int16').tobytes()
            return frame