What is sampling rate?
Sampling is the process of measuring an audio signal at specific points in time. Sampling rate (sr) is a term used in digital audio to describe the rate or frequency at whcih a signal is peridoically samples or measured. The sampling rate or frequency is expressed in Hertz. The sampling frequency in a compact disc player is 44,100 Hertz, which means that the audio signal is sampled 44,100 times per second. The higher the sampling rate, the better the representation of the original analog wave and better the sound quality. See table below.
It is important to prepare the original sound file before re-sampling. For example, if the original sound has background noise, after re-sampling and programmed to the IC, those noises will be increased. Keep in mind that the playback quality can never be 100% as good as the original file because the playback method is not the same. For example, the original sound was played back on the PC with high quality speaker, sound card, etc; while the IC is played back from a circuit directly.
How to calculate
To calculate the maximum duration of each IC, it depends on the memory of the IC. Each of the OTP IC has memory size as below:
The general formula that gives the relation ship between the duration and sampling rate is:
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.
For the same 8M memory IC, if we use 8 bit compression and 12 KHz sampling rate, the duration is reduced from 341 secs to 85 secs.
Sampling Frequency Rate Table (ADPCM / PCM)
Sampling Frequency Rate Table (5-uLaw)
Resistor (ohm) / Rosc values for Sampling Rate
See also PCM vs ADPCM
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