I’m new here and am not sure whether this is the correct place to ask this question. If not, please feel free to direct me to some other proper channels.
I am asked to evaluate a certain IP phone. The voice is obviously garbled, or at least jerky. I browsed the sources and found that the audio driver was initialized with a different sampling rate than the codec sampling rate.
This led me to a query: is it supposed or typical among IP phones that the audio driver be initialized with the same sampling rate required by the codec. If not, are they anywhere related?
This is what I perceive. Correct me when I am wrong. The voice signal from the mic is transformed by the continuous-to-discrete converter with the sampling rate indicated by the audio driver. That is further passed to the codec and through RTP to the remote end. The remote end reconstructs voice from RTP by decoding with the codec sampling rate and finally reach the speaker to be interpolated as a continuous signal.
If my understanding is correct, there are 4 steps subject to the sampling rate. How can we ensure the reconstructed voice is played with the correct frequency?