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Week 2 - Basic Principles

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Forum Index > Classroom 3 - Digital Audio 02 2004


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AUdIoCoUrSeS



Joined: 31 Oct 2002
Posts: 2014
Week 2 - Basic Principles  Reply with quote  

Consider the following and then answer the questions below in your own words:

Why Digital?

Outline of Analogue Signal Processing

• Analogue audio is a mature technology - all of the great breakthroughs in analogue have been made

• Analogue technology is moving towards its limits, governed by the laws of physics.

• The law of diminishing returns applies- as short comings in performance are studied, the equipment needed to counteract them becomes complex and costly.

Weaknesses of Analogue

• Degradation's cannot be separated from the original signal-
at the end of a system a signal carries the sum of all the degradation's introduced at each stage through which it has passed.

• There is a limit to the number of stages through which a signal can be passed before it is useless.

What is Digital Audio?

Digital audio is simply an alternative means of carrying information, although there are a number of ways in which audio can be represented digitally, there is one system, known as Pulse Code Modulation (PCM), which is in universal use.

PCM Key Points

• The waveform is not carried by continuos representations, but by measurement at regular intervals (sampling).

• The frequency with which samples are taken is called the sampling rate or the sampling frequency Fs.

• If every effort is made to rid the sampling clock of Jitter, or time instability, every sample will be made at an exactly even step.

• The length of the sample, which will be proportional to the voltage of the audio waveform, is represented by a whole number (quantizing).

• By describing audio waveforms numerically, the original information has been expressed in a way which is better able to resist unwanted changes.

• The rate at which the voltage is measured (sampled) and the accuracy of the measurement are the only factors which determine the quality, because once a parameter is expressed as a discrete number, a series of such numbers can be conveyed unchanged.

Why Binary?

• Binary has only two digits, 1 and 0.

• With only two states, there is little chance of error.

• These are readily conveyed in switching circuits by an 'off' and an 'on' state.

• In a binary number, the digits represent increasing powers of two from the LSB.

• The bits represent 1,2,4,8,16, etc.

• A multi digit binary number is commonly called a word, and the number of bits in the word is called the wordlength.

• A word of 8 bits is called a Byte.

• The capacity of memories and storage media is measured in bytes.

• The whole number representing the length of the sample is expressed in binary.

• The two states change at predetermined times according to some stable clock.

Further points to note

• The bandwidth of the system effects slew rate (Fourier).

• Noise added to a sloping signal can change the time at which the slicer judges that the level passes through the threshold.

• This effect is eliminated when the o/p of the slicer is re-clocked.

• However many stages the binary signal passes through it still comes out the same, only later.

• Audio samples which are represented by whole numbers can be reliably carried form one place to another by such a scheme, and if the number is correctly received, there has been no loss of information.

Transmission

Two ways binary signals can carry information are:

1. Parallel

2. Serial

• When each digit of the binary number is carried on a separate wire it is a parallel transmission.

• However, using multiple wire is cumbersome.

• A single wire can be used where successive digits from each sample are sent serially. This is the definition of Pulse Code Modulation.

• Clearly the clock frequency must now be higher than the sampling rate.

• A single high quality audio channel requires around 1 million bits per second.

• Data rates needs to be handled economically.

Advantages of Digital Audio compared to Analogue

• The quality of reproduction of a well-engineered digital audio system is independent of the medium and depends only on the quality of the conversion processes.

• The conversion of audio to the digital domain allows tremendous opportunities which were denied to analogue signals.

• Wow, flutter, noise, print-through, drop-outs, intermodulation noise, HF squashing, azimuth error, interchannel phase errors are all history.

• When a digital recording is copied, the same numbers appear on the copy: it is not a dub, it is a clone. If the copy is indistinguishable from the original, there has been no generation loss.

• Digital recordings take up less space than analogue recordings for the same or better quality. Tape costs are far less and storage costs are reduced.

• Digital circuitry costs less to manufacture- switching circuitry which handles binary can be integrated more densely than analogue circuitry. More functionality can be put into the same chip.

• Disk drives and memories developed for computers can be put into use in audio products- there seems to be little point in waiting for a tape to wind when a disk head can access data in milliseconds.

• High performance manipualtion - the difficulty of locating, and the permanence of an analogue edit make it hardly worth considering when the waveform can be viewed on screen, trimmed and auditioned before making it permanent.

• Communication networks developed to handle data can happily carry digital audio over indefinite distances without loss.

• Digital equipment can have self-diagnosis programs built in-
the machine points out its own failures.

• An organisation will still need maintenance staff, but they will be fewer in number and their skills will be oriented more to systems than devices.

• Debates about sound quality are academic; analogue equipment can no longer compete economically.

References and further reading:


1. John Watkinson, "The Art of Digital Audio", Pub. Focal press, 1995

2. Bloom, P.J., "High-quality digital audio in the entertainment industry": an overview of achievements and challenges. IEEE Acoust. Speech Signal Process. Mag., 2, 2-25 (1985)


-------------

1. What are the characteristic of analogue audio?
2. What is digital audio? - include the term PCM.
3. Explain how the binary number system works. Include the terms MSB and LSB along with wordlength.
4. Briefly compare the differences between serial data transfer and parallel data transfer, mention such terms as clock frequency.
5. Outline the history and background of binary code; development of digital audio and the use of binary.
6. Outline the growth of the digital audio industry since the early 80's; - leading bands and pioneering musicians that have exploited the technology, explosion of digital audio.
7. What is quantisation and how many quantising levels are there in a 16 bit digital audio system?
8. What is oversampling?
9. Explain aliasing.
10. Explain the limiting parameters of a hard disk drive that restrict operation in a digital audio workstation.
11. Briefly describe and explain these methods of digital interconnection:

• AES/EBU
• S/PDIF
• SDIF2
• MADI
• SDI and SDTI
• Dolby E
• Proprietary interfaces
• Digital audio via USB
• Digital audio via IEE1394
• Computer systems interfaces
• Other methods of digital interconnection of current relevance
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Post Mon Sep 13, 2004 3:46 am
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iNSTiNCT2765



Joined: 05 Nov 2003
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Location: Denmark
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Week 2 - Basic Principles

(couple questions I couldn't answer) Rolling Eyes

1. What are the characteristics of analogue audio?

Analogue audio is continuous information with an infinite sample rate as opposed to digital audio, where the information is a series of ‘snap shots’. There is generation loss in the signal when transferring between analogue equipment. Analogue tape compression is known to fatten up the audio signal, which is why recording some instruments like drums to tape is still preferred where budget allows it.

2. What is digital audio? - include the term PCM.

Digital audio is an alternative way of carrying information. There is no degradation in the quality when the information is transferred between devices. PCM or Pulse Code Modulation is one of the most common ways of representing digital audio. PCM represents the waveform by measurements at regular intervals also called sampling. The sample rate is the frequency with which these samples are taken.

3. Explain how the binary number system works. Include the terms MSB and LSB along with wordlength.

The binary number system only has two digits, which are 1 and 0. The digits represent increasing powers of two from the right-most bit, which is called the least significant bit (LSB) to the left-most bit, also called the most significant bit (MSB). A binary number that has multiple digits is referred to as a word and the number of bits in the word is called the wordlength.

4. Briefly compare the differences between serial data transfer and parallel data transfer, mention such terms as clock frequency.

Serial data transfer carries information over one single wire where parallel data transfer uses multiple wires so a separate wire is used for each digit of the binary number putting more strain on the system. The clock frequency has to be higher than the set sampling rate for proper functioning.

5. Outline the history and background of binary code; development of digital audio and the use of binary.

? Couldn’t find anything on this.

6. Outline the growth of the digital audio industry since the early 80's; - leading bands and pioneering musicians that have exploited the technology, explosion of digital audio.

Can’t find anything about pioneering musicians and bands but the introduction of the Compact Disc and player would have exploded the growth of digital audio.


7. What is quantisation and how many quantising levels are there in a 16 bit digital audio system?

Quantisation is the process of converting an analogue signal into a digital number. Each bit contains its own quantisation interval therefore a 16-bit digital audio system would have 16 levels of quantisation.

8. What is oversampling?

Oversampling is the process of getting more samples out of an already recorded digital waveform.

9. Explain aliasing.

This is the attempt at trying to record and playback frequencies that are over one-half the sampling rate. This causes the tone to start reflecting after it passes half the sampling frequency, which in turn causes a distortion to the sound.

10. Explain the limiting parameters of a hard disk drive that restrict operation in a digital audio workstation.

Limiting parameters of a hard disk drive include rotation speed, which will determine the speed at which the information on the drive is located. The size of the hard disk drive will also restrict the number of tracks you can record and edit simultaneously as well as the sample rate at which they can be recorded. Higher sample rates take up more disk space so the number of tracks will obviously decrease as the sample rate increases.

11. Briefly describe and explain these methods of digital interconnection:

• AES/EBU – This stands for Audio Engineering Society/ European Broadcasting Union and is a standard for transferring digital audio. The connection is made using one 3-pin XLR connector, which carries both the left and right channel audio.

• S/PDIF – This stands for Sony/Phillips Digital Interface and is a standard for connecting digital equipment such as DAT recorders to other digital equipment such as samplers without any degradation in the sound. The connection can be made with either a RCA connector or an optical connector.

• SDIF2 – This stands for Sony Digital Interface Format and is used on DASH machines.

• MADI – This stands for Multichannel Audio Digital Interface and contains 28 stereo AES/EBU signals in a serial connection.

• SDI and SDTI – This is used to connect standard video devices such as VTRs, switchers, mixers, video servers, video monitors, etc. through a central router. It can carry uncompressed video aswell as compressed video.

• Dolby E – This is a technology that allows up to eight channels of audio to be distributed through any stereo AES/EBU channel.

• Proprietary interfaces – The Roland R-Bus connection which is exclusively used with Roland products transfers 8 channels of audio to and from R-Bus equipped Roland gear or to the RPC-1 PCI card for communication with a computer. The connection is made with a 25 Pin D-sub cable.

• Digital audio via USB – This stands for Universal Serial Bus and is a connection between two pieces of equipment so digital audio can be transferred between them at speeds of up to 480 MB/s using the USB 2.0 protocol.

• Digital audio via IEE1394 – This is the firewire connection between two pieces of equipment and is used to transfer multiple channels of digital audio between them at high speeds over a single cable.

• Computer systems interfaces – PCI cards that can be installed onto the motherboard of a computer can then be interconnected with hardware and breakout boxes at high speeds.

• Other methods of digital interconnection of current relevance - ?
Post Sat Sep 18, 2004 9:44 am
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AUdIoCoUrSeS



Joined: 31 Oct 2002
Posts: 2014
Very Good Start  Reply with quote  

Hey there, a great start!

Just a couple things I want to draw your attention to:



quote:
What is quantisation and how many quantising levels are there in a 16 bit digital audio system?


OK here you need to look at a calculation:

I am asking how many different states of on and off are there if 16 bits are utilisied. i.e. What is the highest number that can be represented with 16 bits?


quote:
What is oversampling?

Oversampling is the process of getting more samples out of an already recorded digital waveform.


Oversampling means using a sampling rate which is greater (generally substantially greater) that the Nyquist rate. Neither sampling theory nor quantizing theory require oversampling to be used to obtain a given singal quality, but Nyquist rate conversion places extremely high demands on component accuracy when a convertor is implemented. Oversamlping allows a given signal quality to be reached without requiring very close tolerance, and therefore expensive components. Although it can be used alone, the advantages of overampling are better realised when it is used in conjuntion with noise shaping. Thus in practise the two processes are generally used together and the terms are often used in the loose sense as if they were synonymous.


The Art of Digital Audio - Watkinson

Benefits:
1. it allows the slope of analog filters to be relaxed
2. allows the resolution of converters to be extended
3. improvements in resolution with noise shaping.
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It's all in the ears. - Learn the concepts not the software. Audio Courses is a way into the music business for you
Post Sat Sep 18, 2004 12:22 pm
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iNSTiNCT2765



Joined: 05 Nov 2003
Posts: 60
Location: Denmark
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Ok, I found out that 16-bit quantisation has 65536 'discrete' levels but I can't find a formula for it or how to get to that number and don't really know why they are discrete levels? Little help please Laughing

- Aman
Post Sun Sep 19, 2004 5:07 pm
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AUdIoCoUrSeS



Joined: 31 Oct 2002
Posts: 2014
Levels  Reply with quote  

OK I want you to dig a bit deeper here, it is important.

Think about binary for the formula, i.e, how is the number 65536 derived, and how many levels for say 20 bits?

Regarding discrete levels, you can think about the amount of quantization steps being available. If a volume control had indents a 16 bit one would have 65536 indents...

Get this?
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It's all in the ears. - Learn the concepts not the software. Audio Courses is a way into the music business for you
Post Mon Sep 20, 2004 7:33 am
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iNSTiNCT2765



Joined: 05 Nov 2003
Posts: 60
Location: Denmark
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Hi Chris,

Ok, the level doubles for every bit added so 1 bit would have 2 levels of quantization and 2 bits would have 4 levels etc. So if you keep doubling, at 16 bits you get 65536 levels of quantization and at 20 bits you get 1048576 levels.

- Aman
Post Thu Sep 23, 2004 5:19 am
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AUdIoCoUrSeS



Joined: 31 Oct 2002
Posts: 2014
Good  Reply with quote  

There we go Smile)

Great stuff!

How's that software?
_________________
It's all in the ears. - Learn the concepts not the software. Audio Courses is a way into the music business for you
Post Thu Sep 23, 2004 5:30 am
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iNSTiNCT2765



Joined: 05 Nov 2003
Posts: 60
Location: Denmark
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Hey Chris,

the software is really good. installation was painless and i worked my way through it pretty quickly. the included samples are of a usable quality and the filters and effects are decent. i'm gonna have another go at it tomorrow probly and really get into it. i'll also try to get a review written..looks like it's gonna be a positive one Wink

- Aman
Post Thu Sep 23, 2004 7:02 am
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