ADPCM (Adaptive Differential Pulse Code Modulation) is a digital audio encoding and compression algorithm used to reduce the amount of data required to represent and transmit audio signals without significant loss in quality. It is a widely adopted method for compressing audio recordings and is often used in telecommunications, speech coding, and multimedia applications.
The ADPCM algorithm works by predicting the difference between consecutive audio samples and encoding this difference instead of encoding the actual samples themselves. This prediction is adaptive and based on previous samples, allowing for efficient compression. The difference between the predicted and actual sample is then quantized and encoded into a binary format, reducing the overall data size.
ADPCM offers a trade-off between sound quality and compression ratio, meaning that higher compression ratios often result in a slight degradation in audio quality. However, the algorithm has been designed to minimize such quality loss, especially when using a suitable sampling rate and bitrate.
One of the key advantages of ADPCM is its ability to encode audio signals with a lower bit rate, making it ideal for applications with limited bandwidth or storage capacities. It enables efficient transmission of audio over narrowband channels, such as telephone lines, as well as digital storage of audio files in various devices.
Overall, ADPCM is a versatile digital audio compression algorithm that efficiently reduces the data size of audio signals while maintaining an acceptable level of audio fidelity, making it an essential tool in the field of audio compression and transmission.
