Sound Theory – Sample Rate, A/D Conversion

It has been a long time since I last posted as I’ve been busy with other jobs. I have recently been refreshing my knowledge of sound theory because I know that it will really help my understanding, especially when it comes to mixing. It’s quite easy to learn all the lingo and tech speak when it comes to sound but not actually, really understand how it works. This is my attempt at explaining it briefly in my own terms.

I understand hertz as being one cycle per second and in digital audio this is used to set a sample rate frequency to convert an analogue source to a digital one. This is performed with my sound card which at the moment is an M-Audio FastTrack USB. The FastTrack is a suitably low-mid range unit which has a guitar and mic level input. Since I just need to plug in the USB to the computer its easy to take the inner workings and process of how the sound is delivered to my speakers for granted. In order to record my voice into the computer I would need to convert the analogue signal of my voice into a digital format for the computer to process. This process is Analogue to Digital Conversion or ADC for short. If the ADC uses a sample rate of 50Hz the analogue signal is split 50 times per second and a measurement of the amplitude taken at that moment in time. This measurement is converted to binary, a measurement known as the bit depth. 50 times per second sounds like a lot but CD quality audio is 44,100 times a second! This value was arrived using the Nyquist Theorem which in a principle I will explain in a later post. When exporting audio I normally choose 44.1Khz as a base amount as my hard drive has Terabytes of space and if I need to convert to MP3 the quality is high enough to start with. By understanding the sample rate and its limitations you can choose lower sample rates like 22Khz or 11Khz. For example a telephone conversation has a bandwidth of 3.6 kHz. “Bandwidth is the difference between the highest and lowest frequencies carried in an audio stream” (Source: Audacity Wiki). Telephone speech sits in this frequency range so 8kHz is a suitable sample rate to select and the file size will be smaller which is useful in some cases like web streaming or computer game audio.

So…the higher the sample rate, the more accurately the computer can reproduce the sound. In order for my ears (analogue system) to hear this played back from the computer the digital audio needs to be converted back to analogue. The sound card performs this using a Digital to Analogue Converter (DAC) which directs the speaker cones to move back and forth at the same frequency of the measured waveform causing pressure changes in the air that my ears convert to brain signals. Amazing when you think about it!

Useful Reading

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