1. Fall 2004EE 3563 Digital Systems Design Audio Basics  Analog to Digital Conversion  Sampling Rate  Quantization  Aliasing  Digital to Analog Conversion

2. Fall 2004EE 3563 Digital Systems Design Audio Basics - Analog to Digital Conversion  Process of digitizing a signal (such as music)  Human hearing range is roughly 20 Hz to 20 KHz  CD’s are sampled at 44,100 Hz – that’s no coincidence  Nyquist Theorem – Must sample at a minimum of twice the highest frequency –If not, undesirable aliasing will occur Music ADCQuantizer LPF Clock – Sampling Rate Samples

3. Fall 2004EE 3563 Digital Systems Design Audio Basics - Analog to Digital Conversion  LPF – Low Pass Filter – used to remove frequencies higher than the Nyquist rate –It’s like turning down the treble on your stereo  Clock is the sampling rate  If clock is 44.1 KHz, then the LPF should remove all frequencies above 22.05 KHz –In practice, you need a little extra removed, so 20 KHz is the cutoff  Sampling rate determines the frequency response –Too low and it will sound like an AM radio –Tradeoff is in data storage space Music ADCQuantizer LPF Clock – Sampling Rate Samples

4. Fall 2004EE 3563 Digital Systems Design Audio Basics - Analog to Digital Conversion  The Analog to Digital Converter gets a sample every time the clock ticks  The sample is passed on to a quantizer  The quantizer outputs a number corresponding to the amplitude of the music at that point  The range of values depends upon how many bits per sample  For CD quality, 16 bits are used (-32768 to +32767)  For voice quality, 8 bits are used (-128 to +127) Music ADCQuantizer LPF Clock – Sampling Rate Samples

5. Fall 2004EE 3563 Digital Systems Design Audio Basics - Analog to Digital Conversion  The fewer the bits, the larger the quantization error, resulting in lower quality  Suppose you gave the teller $93 and asked for change –Suppose she only had twenties (5 steps) – the “quantization” error would be $13 –Suppose she had tens (10 steps) – the error would be $3  The tradeoff is in the amount of data to store  CD Quality: 2 channels * 16 bits/sample * 44100 samples/sec –= 176400 bytes/sec  Voice Quality: 1 channel * 8 bits/sample * 8000 samples/sec –= 8000 bytes/sec Music ADCQuantizer LPF Clock – Sampling Rate Samples

6. Fall 2004EE 3563 Digital Systems Design Audio Basics - Analog to Digital Conversion  The sampling rate and the number of bits/sample together determine the overall fidelity Music ADCQuantizer LPF Clock – Sampling Rate Samples

7. Fall 2004EE 3563 Digital Systems Design Audio Basics - Aliasing  Aliasing occurs when the sampling frequency is below the Nyquist rate  It manifests itself as low frequency noise  Sampling at Nyquist frequency  Sampling below Nyquist frequency

8. Fall 2004EE 3563 Digital Systems Design Audio Basics – Example Waveform

9. Fall 2004EE 3563 Digital Systems Design Audio Basics – Example Waveform

10. Fall 2004EE 3563 Digital Systems Design Audio Basics – Example Waveform 1/5000 th second Note the “squareness”

11. Fall 2004EE 3563 Digital Systems Design Audio Basics – Analog to Digital Converter A basic Analog to Digital Converter (ADC) is shown Note that these comparators are ANALOG comparators The voltage at each point along the ladder drops The comparator output is high when analog input voltage exceeds the reference voltage There is an 8-bit priority encoder internally to produce the digital output

12. Fall 2004EE 3563 Digital Systems Design Audio Basics - Digital to Analog Conversion  For converting back to music, the process is reversed  The LPF is required on the output because it has a staircase shape –In reality, a staircase shape is composed of an infinite number of sine waves of increasing frequency –These frequencies must be removed or the output will be noisy MusicDACLPF Clock – Sampling Rate Samples

13. Fall 2004EE 3563 Digital Systems Design Digital to Analog Converter  There is a resistive ladder that must be very precise  Each of the switches is essentially a mux that switches between the reference voltage and ground  The final output is called a summing amplifier – it is simply an analog adder