I am new to the concepts of Sampling and Quantization, and despite reading many articles, I am not clear on how PCM works if I record a simple sound.
If I record the note A3 (220hz) at 44.1 KHz sampling rate, 16 bit quantization and mono channel for 1 second; from what I read, PCM stream will note the amplitude of the A3 note 44100 times and try to fit the amplitude value in one of the 65536 levels.
While converting this digital signal back to analog, how would we know that the original signal was 220hz ? PCM encoding has only saved the amplitude values - regardless of what frequencies were present in original input.
After reading more, I realized that frequencies are not stored like amplitudes - they are instead calculated using Fourier Transform; or Discrete Fourier Transform for digital signals.