The following projects were judged by the students' peers or by me as being worthy of inclusion on my "Students' Page." They were presented by students in my Speech and/or Communication Media classes as short essays, scripts or productions on a topic within their field of study. Feel free to respond to the individual students through the feedback form below. I will print out your critique and/or comments and return them to the student involved. Please identify the name of the student and/or the title of the entry in the subject listing of the e-mail.
Index: (Sorry, only one at the present time, but we will add others as they become available.)
Anti-aliasing and Sampling
A short essay on audio production
Written by
Angela Adams
Two basic steps used to convert analog signals into digital format are anti-aliasing and sampling. Since it's development in the late 1960's, digital recording has set a new standard in the quality of recorded sound. Unlike analog recording, digital processing offers a much greater dynamic range with a measurable reduction of noise distortion. Improving the quality of analog signals can be achieved by converting them through digital processing.
Analog signals can contain very high frequencies. Although these frequencies exceed the range of human hearing, it is necessary to eliminate them early in the conversion process. This step is known as the anti-aliasing stage. The unwanted high frequencies of analog signals can be transferred to an audible level during digital conversion. This is due to the increased dynamic range available in digital sound. Without anti-aliasing the result would be a significant distortion during recording and playback. The anti-aliasing process requires the use of a device called the low pass filter. The low pass filter is pre-set at a specific range which monitors signals passing through it. If any of the input signals exceed the pre-set range, they are filtered out or eliminated. This step insures high quality sound with a greatly improved signal-to-noise ratio. Anti-aliasing also prepares the signal for the next step called sampling.
In digital recording it is necessary to formulate an accurate representation of the original analog signal. This step of digital conversion is known as sampling. During this process, samples of the original signal, or voltages, are taken at fixed intervals and then coded into pulses which represent the original waveform. The rate at which this sampling occurs is referred to as the sampling frequency. Sampling frequency is significant because if a signal is not sampled enough, there would be insufficient information for decoding. In order for a signal to be successfully encoded at it's highest frequency, it has to be sampled at a rate of at least twice it's frequency. This means that in order for high frequency response in digital recording to reach 20,000 Hz., the sampling frequency must be at least 40,000 Hz. Today digital audio utilizes four sampling rates. Thirty-two kHz is used as the international sampling rate for broadcast digital audio, because the maximum bandwidth in broadcast transmission is 15 kHz. The other three sampling rates of 44.056 kHz., 44.1 kHz. and 48 kHz., are used for disc, compact disc, and digital tape recording.
Even though digital recording involves more cost and must use a number of processes including anti-aliasing and sampling, the end result is high fidelity, low distortion sound. The anti-aliasing step of the digital recording process eliminates unwanted frequencies and provides a greatly improved signal-to-noise ratio. The sampling process, because of it's ability to encode such massive information, provides sound representation that surpasses that of the original signal. Today these digital processes have contributed to a greater realization of the potential of sound quality for the consumer.