Media Devices: Audio. CTEC1465/2018S Computer System Support

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1 Media Devices: Audio CTEC1465/2018S Computer System Support

2 Learning Objective Describe how to implement sound in a PC

3 Introduction The process by which sounds are stored in electronic format on your PC is called sampling. Sampling means capturing the state or quality of a particular sound wave a set number of times each second.

4 Introduction (2) All the characteristics of a particular sound wave amplitude, frequency, timbre need to be recorded and translated into ones and zeros to reproduce that sound accurately within the computer and out to your speakers. Sounds are sampled thousands of times per second. The amount of information stored at each sampling is called the bit depth, and the higher the bit depth, the better the recording.

5 Media: Audio COMMUNICATIONS AND SAMPLING THEORY

6 Baseband Communication Systems Information can be defined in two forms: analog digital Information is represented as a signal.

7 Baseband Communication Systems Analog signals: are considered continuous have an amplitude with any number of values between the signal maximum and minimum (the dynamic range )

8 Baseband Communication Systems Digital signals are represented by bits (zeroes and ones) and binary numbers digital signals are considered discrete Digital devices convert analog signals to digital signals using sampling and quantization.

9 Capacity of a Communications Channel Capacity (C), also known as throughput, is the measurement of a communications channel to carry information and is measured in Bits Per Second or bps

10 Capacity of a Communications Channel (with no noise present) where: C = 2 BW log 2 N BW is the bandwidth of the channel in Hertz N is the number of coding levels used when transmitting the message. For example, if binary transmission is being used, N=2.

11 Capacity of a Communications Channel (with no noise present) - 2 Example: If a channel has a 100 khz bandwidth and 8-level coding is being used, the capacity of the channel is: C = 2(100 khz)log 2 8 = 2(100kHz)(3 bits) = 600 kbps

12 Capacity of a Communications Channel (with noise present) From the above equation, to increase the capacity of a channel we simply increase the number of possible coding levels. Sounds too good to be true? Well, in reality it is too good to be true

13 Capacity of a Communications Channel (with noise present) - 2 This is because, in real-life, no channel is noiseless. All channels are affected by noise and there is no way to completely eliminate. Claude Shannon, a communications engineer in the 1940 s came up with a formula that included the effects of noise on the capacity of a channel.

14 Capacity of a Communications Channel (with noise present) - 3 The formula calculates the maximum capacity of a channel before errors in transmission start to occur: C = BW log 2 (1+S/N) where S = signal power (in watts) used in the channel N = noise power (in watts) present in the channel

15 Capacity of a Communications Channel (with noise present) - 4 Notice that as the noise level (N) increases, the capacity of the channel decreases. Just what you would expect. To increase the capacity of a channel with a fixed amount of noise, either increase its bandwidth (BW) or increase the signal power (S).

16 Nyquist Nyquist determined that the number of independent pulses that could be put through a channel per unit time is limited to twice the bandwidth of the channel: where fp is the pulse frequency (in pulses per second) and B is the bandwidth (in hertz). The quantity 2B later came to be called the Nyquist rate, and transmitting at the limiting pulse rate of 2B pulses per second as signalling at the Nyquist rate.

17 NOISE AND COMMUNICATIONS Where does noise originate in a communication system? 1. Channel 2. Equipment DEFINE: -undesired random variations that interfere with the desired signal and inhibit communication.

18 Signal-to-Noise Ratio Signal-to-noise ratio is defined as the power ratio between a signal (meaningful information) and the background noise (unwanted signal) In decibels the formula can be rewritten as: where A is root mean square (RMS) amplitude (for example, RMS voltage).

19 Signal-to-Noise Ratio (2) Dynamic range measures the ratio between the strongest un-distorted signal on a channel and the minimum discernable signal, which for most purposes is the noise level. SNR measures the ratio between an arbitrary signal level (not necessarily the most powerful signal possible) and noise.

20 Signal-to-Noise Ratio (3) Measuring signal-to-noise ratios requires the selection of a representative or reference signal. In audio engineering, the reference signal is usually a sine wave at a standardized nominal or alignment level, such as 1 khz at +4 dbu (1.228 VRMS).

21 Signal-to-Noise Ratio (4) For digital signals, the number of bits used to represent the measurement determines the maximum possible signal-to-noise ratio. This is because the minimum possible noise level is the error caused by the quantization noise.

22 Signal-to-Noise Ratio (5) For n-bit integers (i.e., fixed point Digital Signal Processing [DSP]) with equal distance between quantization levels the dynamic range (DR) is also determined. The quantization noise is a uniformly-distributed random signal with a peak-to-peak amplitude of one quantization level, making the amplitude ratio 2 n /1.

23 Signal-to-Noise Ratio (6) This relationship is the origin of statements like "16- bit audio has a dynamic range of 96 db". Each extra bit increases the dynamic range by roughly 6 db.

24 References Greg Swick, ELNC545 Course Notes (Winter 2003). John Clark, "CTEC1463 Course Notes" (Winter 2000). Charan Langton, All About Modulation - Part I, Intuitive Guide to Principles of Communications, Retrieved from f on May 22, 2014.

25 References (2)

26 Media: Audio ANALOG-TO-DIGITAL CONVERTER (ADC) & DIGITAL-TO-ANALOG CONVERTER (DAC)

27 Analog-to-Digital Conversion Steven W. Smith, "The Scientist and Engineer's Guide to Digital Signal Processing", 2nd ed. Retrieved from

28 Digital-to-Analog Conversion Steven W. Smith, "The Scientist and Engineer's Guide to Digital Signal Processing", 2nd ed. Retrieved from

29 Digital Signal Processing (DSP) System Steven W. Smith, "The Scientist and Engineer's Guide to Digital Signal Processing", 2nd ed. Retrieved from

30 DSP System The block diagram of a DSP system, as the sampling theorem dictates it should be. Before encountering the analog-to-digital converter, the input signal is processed with an electronic low-pass filter to remove all frequencies above the Nyquist frequency (one-half the sampling rate). Steven W. Smith, "The Scientist and Engineer's Guide to Digital Signal Processing", 2nd ed. Retrieved from

31 DSP System (2) This is done to prevent aliasing during sampling, and is correspondingly called an antialias filter. On the other end, the digitized signal is passed through a digital-to-analog converter and another low-pass filter set to the Nyquist frequency. This output filter is called a reconstruction filter. Steven W. Smith, "The Scientist and Engineer's Guide to Digital Signal Processing", 2nd ed. Retrieved from

32 Human Hearing & Sampling Rate The commonly stated range of human hearing is 20 Hz to 20 khz. An audio CD can represent frequencies up to khz, the Nyquist frequency of the 44.1 khz sample rate.

34 Media: Audio PC AUDIO HARDWARE & SOFTWARE

35 AC 97 Audio codec standard developed by Intel in Up to 6 channels of audio 48/96 khz sample rate 16 or 20 bit surround sound

36 Intel High Definition Audio Audio /high-definition-audio.html Audio codec standard developed by Intel in 2004 to replace AC streams of 16 audio channels 8 to 32 bits 6 to 192 khz sample rates

37 Sound Cards Every sound card supports two speakers or a pair of headphones, but many better sound cards support five or more speakers in discrete channels. These multiple speakers provide surround sound and bass through a subwoofer. Better sound cards have a lower signal-to-noise ratio and support for multiple audio connections, such as a microphone, line in, and S/PDIF.

38 Sound Files The popular WAV file format (as well as most other recorded sound formats) is based on PCM (Pulse Code Modulation uncompressed audio.) WAV files can be huge, especially when sampled at high frequency and depth, so compression is a popular way to reduce the file size of recorded sounds. The most popular compressed file type is MP3. A large number of other sound file formats are available, such as AAC and WMA.

39 Player Software To play sounds, you must have some form of player software, such as Windows Media Player. Not all players can play all types of sound files. Some file formats, such as RealMedia, require their own proprietary players.

40 Links What Does It Take To Turn The PC Into A Hi-Fi Audio Platform? The Scientist and Engineer's Guide to Digital Signal Processing

41 Links How to get better quality audio and music from your PC: everything you need to know about FLAC and DACs html Digital Audio Fundamentals

42 Links ormats mats

Chapter 3 Data and Signals

Chapter 3 Data and Signals 3.2 To be transmitted, data must be transformed to electromagnetic signals. 3-1 ANALOG AND DIGITAL Data can be analog or digital. The term analog data refers to information that