Well, not secrets, but many smart people\u2014people who\u2019ve done DSP programming for years\u2014don\u2019t know these points. They have other beliefs that have served them well, but have left gaps.<\/p>\n
Let\u2019s see why<\/h3>\n
Analog audio, to digital for processing and storage, and back to analog<\/p>\n
Component details\u2014first the analog-to-digital converter (ADC)<\/p>\n The digital-to-analog converter (DAC)<\/p>\n Analog to digital conversion, and conversion back to analog are symmetrical processes\u2014not surprising.<\/p>\n But we can make another important observation: We know that the bandlimiting lowpass filter of the ADC is there as a precaution, to ensure that the source signal is limited to frequencies below half the sample rate. But we have an identical filter at the output of the DAC\u2014why do we need that, after eliminating the higher frequencies at the beginning of the ADC? The answer is that conversion to discrete time adds high frequency components not in the original signal.<\/p>\n Stop and think about this\u2014it\u2019s key to understanding digital audio. It means that the digital audio samples do not represent the spectrum of the bandlimited analog signal\u2014the samples represent the spectrum of the bandlimited analog signal and additional higher frequencies.<\/p>\n To understand the nature of the higher frequencies added in the sampling process, it helps to look at the origins of sampling.<\/p>\n Next: We explore the origins of sampling in Part 2<\/em><\/p>\n","protected":false},"excerpt":{"rendered":" I\u2019ll start by giving away secrets first: Individual digital samples are impulses. Not bandlimited impulses, ideal ones. We know what lies between samples\u2014virtual zero samples. Audio samples don\u2019t represent the source audio. They represent a modulated version of the audio. … Continue reading ![\"\"](\"\/main\/wp-content\/uploads\/2017\/08\/ADA.png\")
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