Sound can be stored in a Aplus voice IC in two compression formats: PCM (Pulse code modulation) and ADPCM (Adaptive Differential Pulse Code Modulation). It is important to understand the differences between the two, so that you can select the right compression format before programming the Aplus Voice IC. In this article we will explain the difference between these two compression formats.
PCM vs ADPCM
Pulse Code Modulation (PCM) and Adaptive Delta Pulse Code Modulation (ADPCM) are subclasses of the Microsoft waveform (.WAV) file format. In PCM, data for .WAV files is stored using linear samples 8 bits per sample, while ADPCM uses deltas between samples at 4 bits per sample.
As the number of bits per sample is less, ADPCM format takes up a half of the disk space, hence, longer duration sound files can be stored. It stores the value differences between two adjacent PCM samples and makes some assumptions that allow data reduction. Because of these assumptions, low frequencies are properly reproduced, but any high frequencies tend to get distorted. The distortion is easily audible in 11 kHz ADPCM files. Using sound files with the same sample rate, if the quality of the sound is important, it is recommended to use PCM format. Likewise, if the duration is important, it is better to use ADPCM format.
SAMPLING RATE vs DURATION
In general, higher the sampling rate gives higher sound quality, but more memory is needed to store the sound file. The general formula to calculate the duration:
Duration (in seconds) =
Note: The compression bits are ADPCM = 4 bits, PCM = 8 bits, 5-uLaw = 5 bits
For an 8M memory IC with 4 bit compression at 6 KHz sampling rate, the duration is:
Based on the sampling rate, the quality of the IC can be improved. If your application requires better quality, save the sound file at a higher sampling rate like 8 KHz or 12 KHz.
See also What is sample rate?
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