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Media Devices: Audio

CTEC1465/2018SComputer System Support

Learning Objective• Describe how to implement sound in a PC

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.

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.

COMMUNICATIONS AND SAMPLING THEORY

Media: Audio

Baseband Communication Systems

• Information can be defined in two forms:

– analog

– digital

• Information is represented as a signal.


• Analog signals:

– are considered continuous

– have an amplitude with any number of values between the signal maximum and minimum (the “dynamic range”)


• 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.

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

Capacity of a Communications Channel (with no noise present)

C = 2�BW�log2Nwhere:

– 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.

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)log28 = 2(100kHz)(3 bits) = 600 kbps

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…

• 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.

Capacity of a Communications Channel (with noise present) - 2

• The formula calculates the maximum capacity of a channel before errors in transmission start to occur:

C = BW�log2(1+S/N)where

S = signal power (in watts) used in the channelN = noise power (in watts) present in the channel


• 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).


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.

Nyquist

NOISE AND COMMUNICATIONS• Where does noise originate in a

communication system?1. Channel2. Equipment

• DEFINE:-undesired random variations that interfere with the desired

signal and inhibit communication.

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).

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.

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).

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.

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 2n/1.

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.

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, 2002. www.complextoreal.com Retrieved from http://people.seas.harvard.edu/~jones/cscie129/papers/modulation_1.pdf on May 22, 2014.

References (2)• http://en.wikipedia.org/wiki/Signal-to-noise_ratio

• http://en.wikipedia.org/wiki/Shannon-Hartley_theorem

• http://en.wikipedia.org/wiki/Decibel

• http://www.electronicshub.org/modulation-and-different-types-of-modulation/

• https://en.wikibooks.org/wiki/Communication_Systems/

ANALOG-TO-DIGITAL CONVERTER (ADC)&DIGITAL-TO-ANALOG CONVERTER (DAC)

Media: Audio

Analog-to-Digital Conversion

Steven W. Smith, "The Scientist and Engineer's Guide to Digital Signal Processing", 2nd ed. Retrieved from http://www.dspguide.com

Digital-to-Analog Conversion


Digital Signal Processing (DSP) System


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).


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 Nyquistfrequency.

• This output filter is called a reconstruction filter.


Human Hearing & Sampling Rate• The commonly stated range of human hearing

is 20 Hz to 20 kHz. https://en.wikipedia.org/wiki/Hearing_range• An audio CD can represent frequencies up to

22.05 kHz, the Nyquist frequency of the 44.1 kHz sample rate. https://en.wikipedia.org/wiki/Compact_Disc_Digital_Audio

• https://en.wikipedia.org/wiki/44,100_Hz

https://commons.wikimedia.org/wiki/File:Perceived_Human_Hearing.png

PC AUDIO HARDWARE & SOFTWARE

Media: Audio

AC ‘97• https://en.wikipedia.org/wiki/AC%2797• Audio codec standard developed by Intel in

1997.• Up to 6 channels of audio• 48/96 kHz sample rate• 16 or 20 bit• surround sound

Intel High Definition Audio• https://en.wikipedia.org/wiki/Intel_High_Definition_

Audio• http://www.intel.com/content/www/us/en/chipsets

/high-definition-audio.html• Audio codec standard developed by Intel in 2004 to

replace AC ‘97.• 15 streams of 16 audio channels• 8 to 32 bits• 6 to 192 kHz sample rates

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.

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.http://www.mp3-history.com/

• A large number of other sound file formats are available, such as AAC and WMA.

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.• https://www.videolan.org/vlc/index.html

Links• http://www.tomshardware.com/reviews/high-

end-pc-audio,3733.htmlWhat Does It Take To Turn The PC Into A Hi-Fi Audio Platform?

• http://www.dspguide.com/pdfbook.htmThe Scientist and Engineer's Guide to Digital Signal Processing

Links• http://www.pcadvisor.co.uk/how-

to/audio/3491386/how-get-better-quality-audio-from-your-pc/How to get better quality audio and music from your PC: everything you need to know about FLAC and DACs

• http://manual.audacityteam.org/man/digital_audio.htmlDigital Audio Fundamentals

Links• https://en.wikipedia.org/wiki/Audio_signal_processing• https://en.wikipedia.org/wiki/Digital_recording• https://en.wikipedia.org/wiki/Codec• https://en.wikipedia.org/wiki/Comparison_of_audio_coding_f

ormats• https://en.wikipedia.org/wiki/List_of_codecs• https://en.wikipedia.org/wiki/Comparison_of_container_for

mats• https://en.wikipedia.org/wiki/Pulse-code_modulation• https://en.wikipedia.org/wiki/MP3

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