The conversion of an analog speech signal to a pulse code modulation (PCM) digital signal involves all of the following steps EXCEPT:

Low-pass filtering (anti-aliasing) is applied before sampling to remove frequency components above the Nyquist limit, preventing aliasing distortion in the digitized output.

Periodic sampling captures the instantaneous analog amplitude at uniform intervals - for telephony, 8,000 samples per second as required by the Nyquist theorem for a 4 kHz voice band.

Quantizing maps each sample to the nearest value in a finite set of discrete levels, converting continuous amplitude values into a digital numeric representation.

Companding (compression followed by expansion) is used in telephony PCM standards (mu-law in North America, A-law in Europe) to allocate more quantization levels to low-amplitude signals and improve signal-to-noise ratio across a wide dynamic range.

Amplitude modulation (AM) is an analog modulation scheme in which a carrier signal's amplitude is varied to encode information, and it exists entirely within the analog domain on a separate carrier wave. PCM conversion is a digitization process that transforms a baseband analog signal into a binary bit stream through filtering, sampling, and quantization - AM is neither a step in that process nor compatible with the goal of producing a digital output.