ADPCM: A High-Quality Audio Compression Algorithm for Low Bit Rates

ADPCM (adaptive differential pulse code modulation) is a type of audio compression algorithm that uses a combination of differential pulse code modulation (DPCM) and adaptive coding to achieve high quality audio at low bit rates.

In DPCM, the difference between successive samples of an audio signal is encoded, rather than the absolute values of the samples themselves. This technique reduces the amount of data required to represent the audio signal, since the differences between consecutive samples are typically much smaller than the absolute values.

ADPCM adds an additional step to the DPCM process, which adapts the quantization step size based on the magnitude of the difference between successive samples. In other words, the algorithm adjusts the precision of the encoding based on the complexity of the audio signal. When the signal is relatively simple, the quantization step size is increased, resulting in a more precise representation of the signal and a higher quality audio output. Conversely, when the signal is complex, the quantization step size is decreased, resulting in a less precise representation of the signal and a lower quality audio output.

The main advantage of ADPCM is that it can achieve high quality audio at low bit rates, making it well-suited for applications where bandwidth is limited, such as video conferencing, voice over IP, and streaming audio. However, the algorithm can be computationally intensive, which may limit its use in real-time applications.