AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Week 2 - Basic Principles |
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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)
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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
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Mon Sep 05, 2005 7:06 pm |
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Polarman
Joined: 24 Jun 2005
Posts: 55
Location: Barbados |
| Basic Principles |
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Hi here comes my answers. I understand now why you said three modules at once is a lot of work  .
1. What are the characteristic of analogue audio?
Sound by itself is an analogue signal.
The full definition of an ANALOGUE SIGNAL:
a) It is defined for any instant of time that means that for any point in the time spectrum there is a value of the signal.
b) It may have an infinite number of different instantaneous values.
Electronic processing of sound requires a conversion of variation of air pressure variation of an electrical signal, normally a voltage or a current. The microphone is as "air-pressure to voltage converter", while the loud-speaker is the "voltage air pressure converter". Microphone and loud-speaker are analogue converters:
Relationship between voltage and air-pressure is linear (in the ideal case).
While an analogue signal can have infinite values compared to a digital signal which have predefined values according to the sample rate (Hz) and quantisation (bits).
The shortcomings of the analogue signal lies in its own nature:
The playback mechanism has no possibility to differentiate distortion or noise from the original signal because any waveform is allowable.
Every copy that is made has more noise than the copied material.
Both the playback and the recording mechanism must have physically contact with the media that means further damage after each pass.
Each component adds distortion and noise in the signal patch
2. What is digital audio? - include the term PCM.
Digital audio is a representation of the analogue signal. There are many ways to represent audio digitally one of is Pulse Code Modulation (PCM).
When an electrical signal is represented by binary numbers, only these voltages are allowed, which can be represented by these numbers. This means, voltages, which have values which lie between two numbers, are not defined and are not possible. Only Discrete values are allowed.
As electrical signals always require a certain time, to change from one value to another, there will be times, when the signal is not defined. So the signal can only be represented at certain, discrete times.
The full definition of a digital signal is:
1. It is defined only for a limited number of values.
2. It is defined only for certain instances.
To convert an analogue signal to a digital signal, two operations are required:
a) The analogue signal must be evaluated at certain instances.
b) The detected analogue values must be converted to binary digital values.
For each of the two operations a certain device will be required.
Each evaluation and conversion of an analogue value, thus each DIGITAL SAMPLE of the analogue signal, will produce a certain amount of data. If a certain time, e.g. 1 second, of a analogue signal is to be converted to a digital signal, the amount of binary data produced depends on:
- The number of samples taken during the period.
- The number of bits used to represent one sample.
The total number of bits produced is just the product of samples and bits per sample.
The dynamic range of a signal describes the range between the smallest signal that can be distinguished from the noise level and the highest signal the system can handle. The smallest signal in a digital system is one bit. Any smaller signal will disappear in the quantization noise. The largest signal the system can represent is given by the highest binary number of the code.
3. Explain how the binary number system works. Include the terms MSB and LSB along with word length.
Binary system uses only the digits 0 and 1. These are named binary digits whereas called BITS. We as humans normally use the decimal system with the base of ten. There are also hexadecimal system with the base of 16, octal system with the base of 8 and so on. In the decimal system we are used to that the digits from the right (LSB) to the left (MSB) means ones, tens, hundreds, thousands and so on. In the binary system counting from the LSB the digits means 1, 2, 4, 8.
The decimal serie:
0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10
his would be in binary, if you are using a 4 bit word length
0000, 0001, 0010, 0011, 0100, 0101, 0110, 0111, 1000, 1001, 1010
4. Briefly compare the differences between serial data transfer and parallel data transfer, mention such terms as clock frequency.
Serial and parallel data transfers are two different ways to send audio samples.
When each wire is carrying each bit in the word its parallel transfer. The state of the wire varies with the sampling rate which means the clock frequency is the same as the sampling rate. It has more complex electronics then serial transfer. Signals on multiple data lines can arrive at the receiving side at different times and must be aligned in order to turn the bits into meaningful bytes.
In serial data transfer only one cable is used send the whole word through the same cable. PCM is serial data transfer. Here the clock frequency must be higher then the sampling rate since it must let the whole word to go through at the sampling rate.
5. Outline the history and background of binary code; development of digital audio and the use of binary.
You can found a lot on internet regarding the origin of binary code. But it seems that the modern binary code as we know it was first introduced by Leibniz around 1670 even if he said that the Chinese had preceded him. Leibniz was unable to get funding for his "digital" mechanical calculator, and his binary code was forgotten until the 20th century. While computer scientists recognized the utility of the binary code, they still regarded it as essentially a mathematical tool. But in 1937, Claude Shannon published a landmark thesis ("A Symbolic Analysis of Relay and Switching circuits"), and gave the binary code a new dimension.
Shannon made two important points. First, he recognized that Leibniz's binary code fitted perfectly with Boolean algebra. The latter, invented in the 19th century by mathematician George Boole, had changed the nature of mathematics, but its usefulness was only fully recognized with the advent of the modern computer. Shannon showed that Leibniz's binary code was perfectly suited for the implementation of Boolean logic in electrical circuits.
Some milestones in digital audio:
1959: Jack Kilby of Texas Instruments patented the first integrated circuit.
1969: Philips began development to record video images in holographic form on disc.
1973: Philips Research begins the development of an audio disc with optical readout.
1976: The first 16-bit digital recording in the US was made at the Santa Fe Opera on a handmade Soundstream digital tape recorder developed by Dr. Thomas G. Stockham.
1977: The first digital audio disc prototypes were shown my Mitsubishi, Sony and Hitachi at the Tokyo Audio Fair.
1977-78: Sony and Philips demonstrate the first prototypes of a digital sound system using a laser disc.
1978: Pulse-code Modulation (PCM) consumer audio recording began in June with the introduction of the PCM-1 14-bit Betamax accessory for recording and playback of digital audio with an 80db dynamic range.
1978: Major companies involved in CD technology and manufacturing realised that a CD standard need to be set. The "Digital Audio Disc Convention" was held in Tokyo, with 35 different manufacturers attending. At the time the discs used in the development were 30cm in diameter. Phillips, who were involved in CD technology proposed a disc which was 11.5cm in diameter.
1979: Sony and Philips join forces to develop a world standard for the CD.
1979-80: Six meeting of Sony and Philips take place at offices in Tokyo and Eindhoven. Two working prototypes are demonstrated, a Sony 30 inch optical disc and an 11.5cm disc by Philips. The result of these meeting were that playing time of just over 74 minutes is agreed and the diameter of the disc is increased to 12cm.
1980: Decision made to adopt the sampling frequency at 44.1 kHz, 16-bit resolution, and the cross interleaved Reed-Solomon code (CIRC) for its error correction system.
1982: Sony CDP-101 was introduced in Tokyo, in Europe in the Fall, and in the U.S. in the Spring of 1983.
1984: Sony introduces the D-5 portable compact disc player at the Japan Audio Fair and in the U.S., using the same VSLI chip and miniature laser from the Sony auto CD Player.
1986: Digital Audio Tape (DAT), introduced by Sony/Philips as a result of effort of the 81 member firm R-DAT consortium to develop a recordable version of the optical compact disc. Due to copyright problems, electronics firms delayed development of sunsumer products and DAT remained a high priced professional medium.
1987: Dolby Pro Logic used low-cost integrated circuit chips to encode surround channel information for home speakers.
1988: Sony introduced the D-88 Pocket Discman capable of playing 3 inch compact discs (CD-singles) introduced in 1987.
1991: The Alesis Corporation of Los Angeles introduced on 18 January at National Association of Music Merchants show its new ADAT machine that recorded 8 tracks of digital audio to a standard S-VHS videocassette
1993: The Tascam division of TEAC introduced in February the DA-88 digital 8-track recorder using the Hi-8 videotape.
1999: Sony introduces its "Sony MS Walkman", a palm size player that downloads music from the Internet onto a "Memory Stick Walkman", while protecting copyrights.
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.
In the late 1970s and early 1980s there was a great deal of innovation around the development of electronic music instruments. Analogue synthesisers largely gave way to digital synthesisers and samplers. Early samplers, like early synthesisers, were large and expensive pieces of gear -- companies like Fairlight and New England Digital sold instruments that cost upwards of $100,000.
In the mid 1980s, this changed with the development of low cost samplers. From the late 1970s onward, much popular music was developed on these machines. Groups like Heaven 17, Severed Heads, The Human League, Yazoo, The Art of Noise, Orchestral Manoeuvres in the Dark, Depeche Mode and New Order developed entirely new ways of making popular music by electronic means. Fad Gadget is cited by some as a father to the use of electronics in New Wave.
The natural ability for music machines to make stochastic, non-harmonic, staticky noises led to a genre of music known as industrial music led by pioneering groups such as Throbbing Gristle (which commenced operation in 1975) Wavestar and Cabaret Voltaire. Some artists, like Nine Inch Nails, KMFDM, and Severed Heads, took some of the adventurous innovations of musique concrète and applied them to mechanical dance beats. Others, such as Test Department, Einstürzende Neubauten, took this new sound at face value and created electronic compositions. Meanwhile, other groups took these harsh sounds and melded them into evocative soundscapes. Still others (Front 242, Skinny Puppy) combined this harshness with the earlier, more pop-oriented sounds, forming electronic body music (EBM).
Allied with the growing interest in electronic and industrial music were artists working in the realm of dub music. Notable in this area was producer Adrian Sherwood whose On-U Sound record label in the 1980s was responsible for integrating the industrial and noise aesthetic with tape and dub production with artists such as the industrial-funk outfit Tackhead, vocalist Mark Stewart and others. This paved the way for much of the 1990s interest in dub, first through bands such as Meat Beat Manifesto and later downtempo and trip hop producers such as Kruder & Dorfmeister.
Recent developments: 1980s to early 2000s
The development of the techno sound in Detroit, Michigan and house music in Chicago, Illinois in the early to late 1980s, and the later UK-based acid house movement of the late 1980s and early 1990s all fuelled the development and acceptance of electronic music into the mainstream and to introduce electronic dance music to nightclubs. The falling price of suitable equipment has meant that popular music has increasingly been made electronically. Artists such as Björk and Moby have further popularized variants of this form of music within the mainstream.
7. What is quantisation and how many quantising levels are there in a 16 bit digital audio system?
If the sampling rate determines how often you take a sample from the sound (waveform) then the quantisation process convert the analogue wave’s amplitude to a whole numerical value. Each sample is rounded to the nearest whole number. The higher word length you work with the more exact can this value be, since you have more values to assign.
The easiest way to explain how you calculate quantising levels is to start with 2 bit word length. A 2 bit word length has 4 quantising levels:
00, 01, 10, 11 (in decimal: 0, 1, 2, 3). A way of calculate this here is to multiply the amount of states each bit can have. Here it would be 2x2. For a 16 bit word this means 2x2x2x2....16 times(2 raise by 16)= 65526 levels, for a 24 bit word 2 raise by 24= 16777216 levels
8. What is over sampling?
Over sampling was one of the biggest improvements to digital audio when it came in the late 80s. Over sampling is used in the A/D and D/A converters and in sample rate converters.
A/D Converters
The sampling rate can be up to 128 the standard Nyquist rate. This used to be able to spread the converters noise above the audible frequency range which results in a cleaner signal. After that is done the signal is digitally down sampled to desired sampling rate and word length. The down sampling is done by something called a decimator which contains a digital low-pass filter at half the sampling rate to eliminate aliases.
D/A Converters
To be able to convert the signal with a higher sampling rate then used. New samples must be inserted between the samples. Normally here the digital filter multiplies the sampling rate 2 to 8 times moving distortion and artefacts above the audible frequency band.
9. Explain aliasing.
According to Nyqvist sampling theorem must the sampling frequency be at least double the sampled signal (sinus or more complex) to be reconstructed.
An example is if you are using a sampling frequency of 44.10 Hz than you can only sample frequencies < 22.05 Hz (included any components in the wave). This limit frequency is often called the folding frequency. The folding frequency is always half the sampling frequency. If you are sampling frequencies that are higher up you get aliasing. When you are reconstructing your sample you will have a lower frequency then the original wave. It’s often good to have a low pass filter on the input.
Good ref. http://www.dsptutor.freeuk.com/aliasing/AD102.html
10. Explain the limiting parameters of a hard disk drive that restrict operation in a digital audio workstation.
Size
Today maybe the size is not a big problem. But still one minute 44.1 kHz/16 bit takes up around 5.3 Mb. If you are recording 24 tracks it will use 5.3x24=127 Mb. If you use higher samplings frequency or bit resolution the amount you have to store rises drastically.
Access time for write/read it’s critical since multitrack recording involves a lot of reading and writing to and from the disk. To reduce the amount of read and write operations its important to defragment your hard after each session.
Noise – If you have your computer in the room you are recording in you have to have very quite hard drives. To cover your drive can build up extra heat.
Storage time
Hard drives are mechanical so to store files for archives it’s not a good idea its better to back on DVD.
Buffer cache
When choosing hard drive for audio recording its good to have as big buffer cache on the hard drive as possible. A cache is basically a local memory for the hard drive. The hard drive stores information in its memory, so that, if the same information is requested again, it can be read from cache memory faster than it would take to re-read it from the disk. Higher capacity improves performance, particularly where the repeat information is constantly being accessed. This is an essential feature for stability and reducing wear and tear due the constant and rigorous demands of audio processing.
11. Briefly describe and explain these methods of digital interconnection:
AES/EBU
AES3 interface (The interface formerly known as AES/EBU). It’s a short term for American Engineering Society/European Broadcasting Union)
AES/EBU is a serial transmission that has become the dominant standard for the interconnection of digital audio signals between equipment - audio and video. The transmission is two channels of digital audio data on a single twisted-pair cable using 3-pin (XLR) connectors. The consumer version is referred to as S/PDIF below.
S/PDIF
S/PDIF is a short term for Sony/Philips Digital InterFace and it is the consumer version of AES/EBU. S/PDIF is a serial interface for transferring digital audio between devices. S/PDIF uses unbalanced 75 ohm coaxial cable up to 10 meters terminated with RCA connectors.
SDIF2
SDIF (Sony digital interface format) Sony’s professional digital audio interface utilizing two BNC-type connectors, one for each audio channel, and a separate BNC-type connector for word synchronization, common to both channels, in all it consists of 3 coaxial cables. All interconnection is done using unbalanced 75 ohm, coaxial cable of the exact same length (to preserve synchronization), and is not intended for long distances.
MADI
MADI (Multichannel Digital Audio Interface) is used on big open-reel digital multitracks and carries up to 56 channels. It is usually connected as a single BNC-terminated cable for audio data with a second for word clock, but optical versions are also available.
SDI and SDTI
Serial Data Transport Interface (SDTI) is the standard for transporting audio, video, and data between cameras, VTRs, editing/compositing systems, video servers, and transmitters in professional and broadcast video environments. SDI has been used to meet this need, but signals must be uncompressed at the output of a device and recompressed at the input of the target device. Repeated compression/ decompression passes degrade the signal unnecessarily. SDTI was built on the SDI base to provide a mechanism for exchanging digital audio and video signals in their native compressed formats.
Dolby E
Dolby E is a compression technology allowing eight channels to be distributed in a AES/EBU signal or in another words Dolby E coding will allow the two PCM audio channels to be replaced with eight encoded audio channels. This is designed for TV broadcast and production professionals.
Proprietary interfaces
ADAT: Alesis’s 8-track digital tape deck. Uses a proprietary optical format also known as Lightpipe.
TDIFF: Tascam Digital Interface File Format. The proprietary format used by Tacsam’s DA-88 digital 8-track tape deck.
Digital audio via USB
There is a wide variety of USB (Universal Serial Bus) audio and MIDI interfaces available for Windows and Macintosh. The original USB spec (USB 1.1) only allows for data throughput speeds of about 1 megabyte per second. Because of this, most USB audio interfaces cannot reliably record multitrack audio into a computer, but most do work well enough for stereo (two channels). USB 2.0 has claimed throughput of over 50MB per second and that makes it possible to use for multitrack recording.
Digital audio via IEE1394
IEEE-1394 (Firewire) A joint Apple and TI implementation of the IEEE P1394 Serial Bus Standard. It is a high-speed (400/800 Mbits/sec) connection faster than USB. There are many audio interfaces using firewire.
Computer systems interfaces
???
Other methods of digital interconnection of current relevance
Examples for digital interconnection could be Ethernet and internet.
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Books
Mastering Audio The art and the science, Bob Katz
Sound and Recording, An introduction Francis Rumsey & TimMcCormick
The Art of Digital Audio, John Watkinson
Internet Sites
http://telecom.tbi.net/history1.html
http://www.tc.umn.edu/~erick205/Papers/paper.html
http://www.aes.org/aeshc/
http://www.pcmag.com/encyclopedia_term/0,2542,t=AESEBU&i=37583,00.asp
http://www.ccru.net/digithype/Afrobinary.htm
http://www.dsptutor.freeuk.com/aliasing/AD102.html
http://www.atpm.com/6.05/digitalaudio.shtml
http://tronweb.super-nova.co.jp/characcodehist.html
http://www.sweetwater.com/creation_station/tech/harddrive.php
http://www.ieee.org/organizations/history_center/sloan/DAR/dar_index.html
http://www.earlevel.com/Digital%20Audio/Oversampling.html
http://www.midiweb.com/info/library/digital-audio.shtml
http://www.tnt-audio.com/clinica/diginterf1_e.html
http://www.dolby.com/professional/pro_audio_engineering/solutions_dolbye.html
http://en.wikipedia.org/wiki/Electronic_music
http://web.media.mit.edu/~joep/SpectrumWeb/SpectrumX.html
http://www.intuitivemusic.com/techno-guide-time-line.html
http://www.classicalworks.com/html/glossary.html
http://www.nvision1.com/Serv/Support/bulletins/digaudio.pdf
http://www.rane.com/pdf/digidic.pdf
http://www.matrox.com/video/press/papers/sdti.cfm
http://www.findarticles.com/p/articles/mi_m0NTN/is_51/ai_112130247
http://psbg.emusician.com/ar/emusic_harddisk_recording/
http://www.soundonsound.com/sos/oct98/articles/digbasics.html
http://www.answers.com/topic/parallel-vs-serial
http://64.233.187.104/search?q=cache:8s0IbjIHAUIJ:www9.dw-world.de/rtc/infotheque/digital_signal/properties.pdf+What+are+the+characteristic+of+analogue+audio&hl=sv
http://history.acusd.edu/gen/recording/notes.html
http://www.rpi.edu/~eglash/eglash.dir/ethnic.dir/r4cyb.dir/r4cybh.htm
http://schoolscience.rice.edu/duker/robots/binarynumber.html
http://history.acusd.edu/gen/recording/digital.html
http://www.digitaltelevision.com/publish/dtvbook/ch5.shtml
http://www.tvtechnology.com/features/audio_notes/f_tc_keeping.shtml
http://www.audioholics.com/techtips/specsformats/upsamplingvsoversampling1.php
http://www.virtualrecordings.com/mp3.htm |
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Sun Sep 11, 2005 12:46 pm |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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That's an excellent start you made there and certainly the way to continue, very well done.
I have picked you on 16bit system it is in fact 65536 and not as you typed, I guess it was a typo error.
Great start!
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Sun Sep 11, 2005 5:00 pm |
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Polarman
Joined: 24 Jun 2005
Posts: 55
Location: Barbados |
| Basic Principles |
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Yeah what a bummer  ! |
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Sun Sep 11, 2005 6:41 pm |
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rachelh
Joined: 16 Jan 2005
Posts: 35
Location: Trinidad WI |
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Better late than never....
1. What are the characteristics of analogue audio?
Analogue audio is recorded by converting continuous variation in sound pressure and converting that into variations in electrical voltage. Analogue audio possesses the characteristics of being continuous that is, the originating sound and the recorded analogue audio maintain a physical and temporal relationship as opposed to digital audio where temporal samples of the originating sound are taken. Because the physical characteristics of analogue audio have a physical direct relationship to the originating sound it makes for easy replay. The problem that arises with the replay system is signal degradation, which occurs because the replay equipment cannot differentiate between the ‘wanted’ and ‘unwanted’ signals- such as distortion, noise, or any other source of interference, which occurred in the recording process. Hence, there is a limit to the usability of an analogue recording, that is, eventually the recording would become useless due to processing the audio over a period of time. Analogue audio has reached its technological peak as most of the scientific breakthroughs associated with this medium have already been made and steps to correct remaining problems have proven not to be cost effective. [1][2][3]
2. What is digital audio? - include the term PCM.
Digital audio refers to the process by which an audio signal is periodically sampled or saved in contrast to analogue- in which the originating sound is recorded in its entirety. Pulse Code Modulation or PCM refers to one of the ways in which audio can be represented digitally which is universally accepted, it is a conversion method in which digital words in a bit stream represent samples of analogue information. The sample rate refers to the number of samples of an analogue signal in 1 second, whilst the sampling time refers to the elapsed time between the sampling period. Because sampling is directly related to a time component, the sampling rate of a system determines its overall bandwidth – a system with a high sample rate is capable of storing more frequencies in the upper limit. [1][3]
3. Explain how the binary number system works. Include the terms MSB and LSB along with wordlength.
The binary or base 2 system is the language by which digital data communicates. ‘This numeric system provides for a fast and efficient means for storing and manipulating digital data. By translating the alphabet, base 10 numbers or other types of information into binary form represented as on/off, voltage/ no voltage, magnetic flux/ no magnetic flux or logical 1 and 0 conditions’ [1]. The wordlength of a digital word also known as its Bit Depth is used to signify the resolution or dynamic accuracy of the digital sample, which has been taken. Digital words representing audio data are commonly 8-bits to 24-bits long, depending upon their application. Increasing the word length more practically allows for more detailed dynamic range and usually translates into higher quality audio. Within the digital word, the MSB or Most Significant Bit is the leftmost digit in any digital word and has the most impact on its mathematical value. The LSB or Least Significant Bit of a digital word is the one when changed has the least effect on the value of the word itself. In any digital word the MSB is the left most digit whilst the LSB is the right most digit. [1][3]
4. Briefly compare the differences between serial data transfer and parallel data transfer, mention such terms as clock frequency.
Serial data transfer refers to the means of data transport where data is transmitted one bit at a time whereas in a parallel data transfer system several bits can be sent at time along multiple wires. Serial data transfer is inherently slower than its parallel counterpart. Clock frequency refers to a unit of measure for the speed of the microprocessor of the computer; the clock cycle is the number of instructions that take place during each cycle that is, the amount of work it can do in each cycle. Hence the clock frequency will have an effect on the speed of data transfer whether serial or parallel.
5. Outline the history and background of binary code; development of digital audio and the use of binary.
The binary system dates back to Pythagoreans in the 6th Century BC. The modern binary number system however was first fully documented by Gottfried Leibniz in the 17th century in his article Explication de l'Arithmétique Binaire. In 1854 mathematician George Boole developed Boolean algebra whose logical system proved instrumental in the development of the binary system, particularly in its incorporation of electric circuitry. In 1937 Claude Shannon founded practical digital circuit design by implementing Boolean algebra and binary arithmetic using electronic relays and switches for the first time in history.
The binary or base 2 system is the language by which digital data communicates. ‘This numeric system provides for a fast and efficient means for storing and manipulating digital data. By translating the alphabet, base 10 numbers or other types of information into binary form represented as on/off, voltage/ no voltage, magnetic flux/ no magnetic flux or logical 1 and 0 conditions’ The wordlength of a digital word also known as its Bit Depth is used to signify the resolution or dynamic accuracy of the digital sample, which has been taken. Digital words representing audio data are commonly 8-bits to 24-bits long, depending upon their application. Increasing the word length more practically allows for more detailed dynamic range and usually translates into higher quality audio. Hence, the binary system is fundamental in digital audio as it is through this system that digital audio exists. [1][3][10]
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.
The growth of the digital audio industry since the early 80’s is due largely in part to the fact that digital is a much cost effective medium than analogue and during that time period there was sort of a technological revolution. Technological advances in computers, processors, as well as the further digitising of musical instruments through synthesisers served to make digital an easily accessible medium that was as effective as its analogue counterpart and the instruments that were digitised. The 80’s spawned musical genres such as new wave that relied highly on synthesisers, which in turn ushered in the dance carve which emerged in the 90’s through techno and more lately trance.
7. What is quantisation and how many quantising levels are there in a 16 bit digital audio system?
Quantization is the ‘division of a continuous event [such as an analogue signal] in to a series of discrete steps’ [3]. In digital audio, quantization takes the form of sampling, where the quantization value represents the amplitude component of the digital sampling process. Quantization itself is the process of translating instantaneous voltage levels of a continuous analogue signal into discrete sets of binary digits for the purpose of storing and manipulating the resultant in the digital domain. In a 16 bit digital audio system the number of quantising levels would be 65,536. [1][3]
8. What is oversampling?
Sampling is the process in the digital domain by which an audio signal is ‘captured’ or ‘stored’ and stored in a binary form. These samples when played back can reconstruct the original audio source. Oversampling is the process by which the sampling rate within a converter’s filtering block is multiplied usually by a range between 12 and 128 times its original rate. This is done to improve anti-aliasing filter characteristics – it has the effect of further reducing intermodulation and other forms of distortion. [1]
9. Explain aliasing.
The Nyquist Theorem states that in order to digitally encode the desired frequency bandwith, the selected sample rate should be at least twice as high as the highest frequency to be recorded. If frequencies greater than half of the sample rate are allowed to enter into the conversion process this results in erroneous frequencies called alias frequencies. Aliasing can be heard as audible harmonic distortion on playback. To combat aliasing a low-pass filter placed before analogue to digital conversion can be implemented. [3][1]
10. Explain the limiting parameters of a hard disk drive that restrict operation in a digital audio workstation.
The limiting parameters of a hard disk drive that restrict operation in a digital audio workstation are:
i. Disk space – determines the number of tracks which can be stored on the disk and also affects the speed of data retrieval.
ii. Disk speed – determines the time taken by the disk head to access data on the disk and is determined by the number of revolutions per minute [rpm] made by the disk
iii. Device latency – the time taken for the device to process a request, the higher the device latency the longer it would take to complete the task at hand
iv. Fragmentation level- this factor slows down data retrieval if the level of fragmentation is high
v. Processor speed- the higher the clock speed of the processor the faster instructions can be executed.
vi. Memory – Cache memory can be essential in DAW as it can house recently edited clips that were placed on the ‘clip board’ as it would not take long to retrieve the information thus increasing the options available to DAW users
11. Briefly describe and explain these methods of digital interconnection:
• AES/EBU
AES/ EBU [Audio Engineering Society / European Broadcasting Union] protocol has been developed has been developed for the transmittal of digital data between digital audio devices. This standard is use to convey two channels of interleaved digital audio through a 3 pin XLR microphone cable. Pin 1 connects to the signal ground whilst pins 2 and 3 are used to carry signal data. This transmission standard is of low impedance about 110 ohms, digital audio channel data and sub-code information is transmitted in blacks of 192 bits that are organised into 24 words, each 8 bits long. [1]
• S/PDIF
S/PDIF (Sony/Philips Digital Interface Format) is considered a consumer format, and is largely based on the AES/EBU standard. In fact, in many cases the two are compatible. There are, however differences between the two formats, particularly in the channel status and user bits. S/PDIF typically uses either unbalanced, high impedance coaxial cables or fibre optic cables for transmission. [3]
• SDIF2
SDIF –2 or Sony Digital Interface Format is the first multi-channel format, it is used in Sony professional digital audio equipment it supports up to 20-bit sampling. For two channel operation, each channel is sent on a separate 75 Ohm unbalanced coaxial cables with BNC connectors all cables should be of the same length. For multi-channel operation, transmits 24 channels on 2 separate balanced cables with multi-pin connectors. A separate cable is necessary with BNC connections for word clock. A master clock - a dedicated master synchronization signal must be applied to all transmitters and receivers. [4]
• MADI
MADI - multichannel audio digital interface. MADI is an extension of the AES3 format (AES/EBU). The standard provides for 56 simultaneous digital audio channels which are conveyed point-to-point on a single coaxial cable fitted with BNC connectors along with a separate synchronization signal. MADI supports up to 24 bit/48 kHz sampling rate although higher rates are possible. Cable length is limited to 5o metres, if longer lengths are necessary fibre optic cables can be used. Data transmission is asynchronously at 100 Mbps. A master clock - a dedicated master synchronization signal must be applied to all transmitters and receivers. [4] [5] [6]
• SDI and SDTI
Serial digital interface [SDI] standard is based on a 270 Mbps transfer rate, over a single 75-ohm coaxial cable [BNC connector], up to 600 feet but shorter lengths are preferred. SDI is used in Television stations, cable channels, and professional production Equipment. SDI provides a method for transmitting uncompressed digital video, audio and other data between video devices
The SDI signal starts with a Timing Reference Signal (TRS) which is a 4 word sequence which contains the unique bit pattern 3FF, 000, 000. The last word is called the Timing Reference Word (TRW), which contains information such as start/end of active video, and the current field. The last digits of the last word are a Hamming Code to protect the TRW. A Hamming Code is a error recovery code, meaning that if any of the bits of the TRW are corrupted, the Hamming code can be used to detect the error and recover the data.
SDTI (Serial Digital Transport Interface) is an expanded specification, which allows compressed (i.e. DV, MPEG and others) video streams to be transported over a SDI line. This allows for multiple video streams in one cable or faster-than-real-time (2x, 4x,...) video transmission. [7] [8] [9]
• Dolby E
Dolby E is a multi-channel coding system developed by Dolby and is primarily used within broadcasting and post- production. Up to eight channels of high-quality audio plus Dolby Digital metadata can be distributed via an AES3 pair, or recorded onto two audio tracks of a digital VTR (Video Tape Recorder). Audio never reaches the consumer in Dolby E form; it is encoded with Dolby Digital just prior to final transmission. To help differentiate their functions, Dolby E is referred to as a distribution coding system, and Dolby Digital as an emission coding system. Dolby E encodes up to eight audio channels plus metadata into a two-channel bitstream with a standard data rate of 1.92 Mbits/sec (20-bit audio at 48 kHz). [7] [4]
• Proprietary interfaces
Proprietary interfaces are those that the manufacturer makes to exclusively work with their product or products from the same manufacturer. Proprietary designs and options allow a manufacturer to implement features that might not be possible if everything were standardized and generic. An example of a proprietary interface would be ADAT Optical (ADI).
• Digital audio via USB
The term USB [Universal Serial Bus] refers to an eternal bus that supports transfer rates of 12 Mbps up to 1.5 Mpbs. USB 2.0 is referred to as Hi-Speed USB it is the most recent version of USB specifications. It is an external bus meaning that it connects from the motherboard to an external peripherals, it supports data rates up to 480 Mbps. USB 2.0 is an extension of USB 1.1. USB 2.0 is fully compatible with USB 1.1 and uses the same cables and connectors. A single USB port can hold up to 127 devices peripherals at any one time. Making USB and USB 2.0 as well as High Speed USB a very convenient and efficient way to connect devices such as printers, scanners, USB MIDI port, cameras etc. to a computer.
• Digital audio via IEE1394
IEE 1934 or Firewire is a "serial SCSI" because it is a serial protocol and conforms to SCSI standards as well. FireWire is fast: it starts at 100 Megabits per second and goes on up past 400 Mbs FireWire is a hot swappable technology and allows 63 devices on a buss with auto termination and identification.
• Computer systems interfaces
The SCSI or [Small Computer Systems Interface], allows many devices (typically up to seven) to connect to one interface card and from there to the motherboard. There has been advances of this interface over the years with each version of course being more advanced than the next in incorporating a newer/ revised version of the standard SCSI cable and connector.
SCSI I
This is the initial SCSI standard. It has a 25 pin connector which can be classified into either SUB –D or Centronics style.
SCSI II
The advent of the SCSI II interface brought about the standard increase in the width of the connector cable to 50 lines as well as the pin being increased from a 25 pin connector to a 50 pin low-density centronics connector and the 50 pin high-density SUB-D connector.
SCSI III
This is the most recent advance in the SCSI chain of evolution, once again the data paths have been expanded, now it resounds at 68 lines and has introduced the high-density 68 pin SUB-D style of connector. This interface provides for fast data transfers and is used exclusively for SCSI fixed disks.
Firewire/mLAN
mLAN refers to a now standard interfacing, which carries; digital audio, MIDI, timecode and drive communications using a singe cable. Firewire refers to a bi-directional interface allowing a single cable to carry MIDI, digital audio or any other presiding signals in both directions.
• Other methods of digital interconnection of current relevance
1. ADAT Optical (ADI) also known as lightpipe supports a 24 bit/48 kHz and sampling rate and transmits 8 channels serially on fibre-optic cable with proprietary connectors. This proprietary interface is used on Alesis ADAT MDM and digital devices such as mixers, synths and effects devices. It is self-clocking although some devices may require a separate 9-pin sync cable to work. [4]
2. TDIF (Tascam Digital Interface Format) is made for Tascam's family of DA-88 recorders and other digital devices such as mixers. This bi-directional interface supports up to 24 bit/multiple sampling rates and transmits 8 channels on multiwire, unbalanced cables with 25-pin D-sub connectors, maximum cable length being 5 meters. Data transmission at 48 kHz sampling rate is 3 Mbps (like AES/EBU). A master clock is necessary however self- clocking can be achieved if LRCK is supported
3. ProDigi
4. Yamaha Y2
=======================
REFERENCES:
[1] MODERN RECORDING TECHNIQUES- DM HUBER, R RUNSTEIN
[2] SOUND AND RECORDING; AN INTRODUCTION –F RUMSEY, T MC CORMICK
[3] WWW.SWEETWATER.COM - IN SYNC
[4] http://www.mtsu.edu/~djbrown/connect.html
[5] http://www.showorks.com/glossary/showglosm.htm
[6] http://www.aes.org/standards/b_reports/news.cfm
[7] http://www.hitachi-service.net/SDI---.pdf
[8] http://www.interfacebus.com/Design_Connector_Video.html
[9] http://www.answers.com/topic/serial-digital-interface
[10] From Wikipedia, the free encyclopedia. |
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Thu Sep 15, 2005 4:09 pm |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Sharing Resources |
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Great stuff Rachel well done, also a good start!
Good to have two of you here now also, and perhaps you will help each other out and share resources.
In fact I think your islands are close to each other geographically...
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Fri Sep 16, 2005 6:28 am |
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rachelh
Joined: 16 Jan 2005
Posts: 35
Location: Trinidad WI |
yep Trinidad and Barbados are really close to each other and I don't mind sharing resources, happy that I am not the ony one in the course...  |
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Fri Sep 16, 2005 7:45 pm |
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AUdIoCoUrSeS
Joined: 31 Oct 2002
Posts: 2014
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| Good |
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Don't forget the classical practical assignment week 4 also applies to you both.
_________________
It's all in the ears. - Learn the concepts not the software.
Audio Courses is a way into the music business for you
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Mon Sep 19, 2005 11:24 am |
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All times are GMT. The time now is Fri May 16, 2008 4:01 pm
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