Pulse-code modulation

Common sample depths for LPCM are 8, 16, 20 or 24 bits per sample.

LPCM encodes a single sound channel. Support for multichannel audio depends on file format and relies on synchronization of multiple LPCM streams. While two channels (stereo) is the most common format, systems can support up to 8 audio channels (7.1 surround) or more.

Common sampling frequencies are 48 kHz as used with DVD format videos, or 44.1 kHz as used in CDs. Sampling frequencies of 96 kHz or 192 kHz can be used on some equipment, but the benefits have been debated.

The Nyquist–Shannon sampling theorem shows PCM devices can operate without introducing distortions within their designed frequency bands if they provide a sampling frequency at least twice that of the highest frequency contained in the input signal. For example, in telephony, the usable voice frequency band ranges from approximately 300 to 3400 Hz. For effective reconstruction of the voice signal, telephony applications therefore typically use an 8000 Hz sampling frequency, which is more than twice the highest usable voice frequency.

Regardless, there are potential sources of impairment implicit in any PCM system:

• Choosing a discrete value that is near but not exactly at the analog signal level for each sample leads to quantization error. When dithering is used to compensate for this, it introduces additional noise.
• Between samples, no measurement of the signal is made; the sampling theorem guarantees non-ambiguous representation and recovery of the signal only if it has no energy at frequency fs/2 or higher (one half the sampling frequency, known as the Nyquist frequency); higher frequencies will not be correctly represented or recovered and add aliasing distortion to the signal below the Nyquist frequency.
• As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or decoding clock is not stable, these imperfections will directly affect the output quality of the device.

Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog domain as part of the analog-to-digital process; newer implementations do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques, such as modified discrete cosine transform (MDCT) coding.

• Linear PCM (LPCM) is PCM with linear quantization.
• Differential PCM (DPCM) encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM.
• Adaptive differential pulse-code modulation (ADPCM) is a variant of DPCM that varies the size of the quantization step to allow further reduction of the required bandwidth for a given signal-to-noise ratio.
• Delta modulation is a form of DPCM that uses one bit per sample to indicate whether the signal is increasing or decreasing compared to the previous sample.

In telephony, a standard audio signal for a single phone call is encoded as 8,000 samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12- or 13-bit linear PCM sample number is mapped into an 8-bit value. This system is described by international standard G.711.

Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8-bit μ-law or A-law PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard.

Audio coding formats and audio codecs have been developed to achieve further compression. Some of these techniques have been standardized and patented. Advanced compression techniques, such as modified discrete cosine transform (MDCT) and linear predictive coding (LPC), are now widely used in mobile phones, voice over IP (VoIP) and streaming media.