A bit (pun intended) about sampling signals via computer

John J. Ohala 2004 (unless otherwise noted)

1. Digital recording eliminates many of the problems associated with analog recording. But with most digital recordings, there is almost always some analog equipment.

For example in the digitization of speech, in most cases the microphone, at least, is an analog device. (In one way or another, the impinging air pressure variations are transduced into voltages. It is the input voltages that are digitized.

John J. Ohala 2004 (unless otherwise noted)

2. The process of digitization is called (as you probably know) A/D (A to D) conversion. The reverse process, where the digitized signal is converted into an analog voltage is called D/A (D to A); this is how digitized audio signals can drive analog equipment like headphones, loudspeakers and the like.
John J. Ohala 2004 (unless otherwise noted)

The following from:

http://www.mathworks.com/access/helpdesk/help/toolbox/daq/c1_int18.shtml

John J. Ohala 2004 (unless otherwise noted)

At insufficient sampling rate, more than one sine wave could end up looking like another of a lower frequency.

John J. Ohala 2004 (unless otherwise noted)

Even though the samples appear to represent a sine wave with a frequency of one-fourth the sampling rate, the actual signal could be any sine wave with a frequency of (n .25) (Sampling rate) where n is zero or any positive integer. For this example, the actual signal could be at a frequency of 3 Hz, 5 Hz, 7 Hz, 9 Hz, and so on. The relationship 0.25 x (Sampling rate) is called the alias of a signal that may be at another frequency. In other words, aliasing occurs when one frequency assumes the identity of another frequency.
John J. Ohala 2004 (unless otherwise noted)

If you sample the input signal at least twice as fast as the highest frequency component, then that signal might be uniquely characterized, but this rate would not mimic the waveform very closely. As shown on the next slide, to get an accurate picture of the waveform, you need a sampling rate of roughly 10 to 20 times the highest frequency.

John J. Ohala 2004 (unless otherwise noted)

John J. Ohala 2004 (unless otherwise noted)

How Can Aliasing be Eliminated? The primary considerations involved in antialiasing are the sampling rate of the A/D converter and the frequencies present in the sampled data. To eliminate aliasing, you must Establish the useful bandwidth of the measurement. Select a sensor with sufficient bandwidth. Select a low-pass anti-aliasing analog filter that can eliminate all frequencies exceeding this bandwidth. Sample the data at a rate at least twice that of the filter's upper cutoff frequency. As a practical measure, it assumed that useful information is obtained at a frequency that is roughly half the sampling rate.
John J. Ohala 2004 (unless otherwise noted)

Some heuristics about sampling rate: 44,100 Hz Industry standard to accommodate full range of human hearing and, especially, the range of frequencies found in music; useful upper range: 22,050 Hz. 22,050 Hz or 16,000 Hz: Good enough for most speech signals; effective useful upper range: 11,025 and 8,000 Hz, respectively. (However, there may be energy in speech, esp. in palatals, up to 16,000 Hz. But it not clear how useful this is to listeners.

10,000 Hz: Adequate for most speech samples (but not necessarily for [s] or [ts]
8,000 Hz: Marginally adequate for speech; covers all important vowel and approximant resonances; close to telephone bandwidth; effective upper range: 4,000 Hz. Potential downside: reduces the individuality of the speakers voice.
John J. Ohala 2004 (unless otherwise noted)

It is important to keep in mind that upper range of the effective bandwidth one gets with a given sampling rate doesnt say anything about the lower frequency limit. Most consumer-grade A/D conversion does not go down to DC (Direct Current = 0 Hz (steady state signals). Many phonetic applications, esp., sampling aerodynamic parameters and other physiological parameters require bandwidth that goes down to DC. In this case one has to invest in a special type of A/D (e.g., from Glottal Enterprises and Scicon) (or other special-purpose A/D designed for the scientific community).

John J. Ohala 2004 (unless otherwise noted)

The problem of quantization: how many bits do you want to devote to measuring the amplitude of the signal.
If you devote only 3 bits (= 8 levels) to quantization

From: http://cnx.rice.edu/content/m0051/latest/
John J. Ohala 2004 (unless otherwise noted)

Today, most consumer-grade quantization goes up to 16 bits (65,536) different amplitude levels. Enough fidelity for most purposes.

Even 12 bit quantization (4,096 amplitude levels) is not bad for speech.

Remember one bit of the number of bits devoted to quantization is the sign bit (+ or -). You have to have this for audio signals; you may not need the negative bits for some physiological signals and you can gain extra quantization by shifting the baseline down, e.g., in the case of 16 bit quantization, devoting only 2 bits to signals below baseline.
John J. Ohala 2004 (unless otherwise noted)

Demo of Praat

John J. Ohala 2004 (unless otherwise noted)