One of the most controversial trends in sound engineering is the so-called loudness war. This name describes the practice of parallel compression that makes the record louder, but at the expense of the dynamic range. This practice results in the loss of sound detail and does not utilize the full potential of the allowed bitrate. This paper will cover the reasons behind the loudness war, whether these records would have to be remastered in the future, and how streaming media has affected this phenomenon.
Discussion
One of the primary reasons behind this practice is the desire of the music labels to cater to the radio market. This loudness is the result of compression done to normalize the sound, making quiet moments of the record as loud as all the other ones. This normalization has a perceived benefit for the radio stations because, in theory, it would prevent listeners from having to adjust the volume on their radios. This is especially important in the case of people who listen to the radio while driving. No distraction should be permitted, so the radio stations tend to prefer records that have gone through this procedure (Seagull par. 5). Another reason lies in the environment that the music is often listened to today. High dynamic range had a clear benefit when people could only listen to music at home and on a high-quality machine. It is much harder to hear the benefit of it while driving a loud car. The noise in the cabin drowns out the quiet parts of the record, removing the benefit of the high dynamic range for the driver (Jones par. 12).
Many engineers point out that this is not a new trend, it has existed since the beginning of music mastering. However, in the recent years, it has become a clear problem. The records are much louder than before and are getting louder at a much faster pace (Jones par. 5). The start of this dramatic increase is often placed at the 1990s. With the popularity of Compact Disks, high-quality digital recordings became available for people to listen in their cars and on the go. This availability led to a desire to make certain records stand out by increasing their volume. In turn, this move created a perception that loud records are more attractive to the modern listener, and with the popularity of noisy styles of music like Grunge, it seemed like not a lot would be lost by applying this practice. Unfortunately, this led to the further increase in compression among other records too. Loudness became competitive. However, all evidence points to the fact that loudness does not affect the purchasing decisions of the customers, so the benefit of this is very questionable.
Modern portable sound systems are much more capable of playing high-quality music, than those from the 1990s. This fact slowly brings attention to the issues of the loudness war. Concerned listeners are organizing petitions and movements to inform the music industry about their preferences, and some artists support them. Some albums, however, would have to be completely remastered in the future. Even if loudness stops being a trend, albums released during this period would have to be remastered to gain the full dynamic range, intended by the musician. A similar issue is seen in the popularity of music streaming services. They often use an improper sound normalization. It prevents the albums from sounding like they were intended, and many users have already expressed concern about this issue ( Dynamic Range Day ).
Conclusion
High dynamic range is important for the complete music experience. The loudness war is making it difficult to gain this complete experience, even in the home environment. Hopefully, in the future, it will become less of a concern.
Works Cited
Dynamic Range Day. Mastering Media Ltd, 2017, Web.
The history of sound recording and systems that were involved in this process is rather interesting, as it is hard to imagine that the first attempt to record a sound was simple and genius simultaneously. The history itself is closely linked with the development of technologies and formats of sound recording, hence, this paper aims to review the key stages of sound recording technology, as well as sound recording equipment and formats.
The Dawn of Sound Recording
Let alone barrel organs, the principle of which was invented in the IX century, the first attempts of acoustic sound recording were taken in 1857 by Édouard-Léon Scott de Martinville who invented the phonautograph. This looked like a socket-pire with a membrane, and a pen attached to the membrane. The membrane vibrated, shaken by the sound, and the pen registered this vibration on the paper roll. The key disadvantage of this method is that the sound could not be played back immediately. However, when phonautograph records were photo engraved on metal plates, the attempt to playback the sound was successful. (Fowler, 1967)
The idea of phonautograph was used by Thomas Edison for his phonograph. However, instead of paper or photo-plate, he used more impressionable materials like tin, wax, or lead, and a stylus instead of a pen to make the groove. The playback was performed by tracing the stylus along the groove. Hence, the vibration was made by the varying depth of the groove, while the system of levers made the sound membrane vibrate and make the sound. However, the barrels could not be copied easily, and this was the key difficulty associated with storing the recorded information. The principle of impressionable material recording was used for gramophone and vinyl disks that became a whole epoch in sound recording, and even the early tape records were of lesser quality in comparison with properly recorded vinyl sound. (Borwick, 1996)
Electric Sound
This technology is close to leaving a groove on a plate or a barrel, however, the groove was replaced by a magnetic signal which was left on a sensitive material. First, this principle was used by Valdemar Poulsen who used it in his telegraphone. The magnetic sensitive material is used as a medium, and sound is recorded by a magnetic head. When the sound needs to be played back, the tape is let through the magnetic sensitive head, which interprets the magnetic signal into the electric one. The first mediums that were used are the wires which were able to preserve the magnetic signal for a long time. (Blanning, 2008)
The next step of electric sound evolution is the invention of tape recording. The older generation remembers tape reels with ¼ recording tape which was widely used either for music playback or educational aims. The tapes were first invented by the engineers of AEG, and the key principle of sound quality improvement was the biasing of the signal within the inaudible sound band ranging from 50 to 150 kHz. This technology was used for a long time and endured numerous improvements. The key and the most important improvement was the invention of multichannel recording when a tape was divided into several lines, and each line included a separate signal. Multitrack audio recording started its history in 1943 when 2 channel tapes and recorders appeared. The professional recording systems could record up to 24 channels, however, the audiophiles adored quadraphonic records.
Audiocassettes were the next evolutionary step of audio format, though, the principle stayed the same. This helped to decrease the weight of a sound recording, however, the actual quality of the sound was sometimes lower in comparison with audiotapes. (Koetsier, 2001)
Digital Recording
The first steps of audio recording with digital sound were realized by using the well-known audiocassette. The form-factor of the cassette was changed for the specialized players which were able to play back the digital sound. It was not widely used by consumers, as it was too expensive, though, radio stations and studios preferred them for a comparatively higher quality of sound. The magnetic head read the digital signal, and then the processing block created the analogous signal needed for the loudspeakers. This principle of digital audio recording was used for creating the PC formats. The binary signal which is the basis of digital recording became the following step of sound recording, storage, and playback, while the evolution baton was picked up by media carriers and algorithms of digital signal processing. Hence, various lossy and lossless formats appeared, and the data carriers develop independently on the sound recording evolution.
Conclusion
Sound recording and playback have reached the essential evolutionary heights, while the first attempts will stay only in the memories of historians and as the museum expositions. Sound recording evolution ranges from the most ancient sound barrels used for church organs, and to the latest digital recording and digital signal processing algorithms. Though, every epoch is featured with unique and interesting media that was used for recording and playback.
Reference List
Blanning, T. (2008). Facing the Music; throughout History, Musicians and Composers Have Battled Rampant Piracy Unscrupulous Publishers and Dubious Employment Practices. The Problems of Todays Recording Industry Pale in Comparison, Writes Time Blanning. New Statesman, 137, 40.
Borwick, J. (Ed.). (1996). Sound Recording Practice (4th ed.). Oxford: Oxford University Press.
Fowler, Charles B. (1967) The Museum of Music: A History of Mechanical Instruments, Music Educators Journal. The National Association for Music Education 54 (2): 4549
Koetsier, Teun (2001). On the prehistory of programmable machines: musical automata, looms, calculators , Mechanism and Machine theory 36, pp. 590591.
James Jordan, in his book Evoking Sounds, Fundamentals of Choral Conducting 2nd edition, gives an overview of how music creators can allow their music to speak to the right audiences. In chapter 23, the author explains that there are four main phases that have to be considered when creating memorable music. The four phases are preparing and marking the score, arranging the tonal materials of the music, anticipating tonal problems, and working on the text. Jordan argues that the four phases have to be followed in a sequence in order for music to resonate with both the singer and the audience.
The first phase, preparing and marking the score, has nine factors to be considered. First, one has to hum through the strings and then sing all the parts of the song.1. Jordan explains that the third step is to practice Alexander-based alignment sensations while staying aware of body mapping principles.2. The fourth step is marking the score, while the fifth is conducting while humming the piece.3. Jordan goes further and explains the importance of studying the breath of the music as the sixth step and then performing the piece while inhaling and exhaling frequently.4. The eighth step is breathing the color and effect of the music, and lastly, one has to study the breath process that connects the phases for a full sound in the ninth step.
Additionally, the author explains that preparing tonal materials for the music allows the composer to learn the tonal aspects of the score.5. This also makes it easier for the composer to then anticipate any vocal problems, which can be solved by ensuring singers have the ability to perform the highest and lowest pitches of the piece. Practices such as correct posture, flexibility, and expanding the tract are all useful to enhance the ability of the singer to perform the piece. According to the scholar, the last step is to prepare the text, which will ensure the composer understands the color of the music.6.
Bibliography
Jordan, James. Evoking Sounds, Fundamentals of Choral Conducting. 2nd ed. Chicago: GIA Publications, 2009.
Footnotes
James Jordan. Evoking Sounds, Fundamentals of Choral Conducting, 2nd ed. (Chicago: GIA Publications, 2009), 307.
Today, the soundtrack is an integral part of the movie industry. It is difficult to imagine watching a film without listening to the dialogues between characters. Even in movies where conversations are kept to a minimum, various sounds are used to set the atmosphere of the scene. This report will outline three facts about the use of sound in film, as discussed in Making Waves: The Art of Cinematic Sound.
Making Waves reveals Edisons motion picture camera and was intended as a direct follow-up to his invention of the phonograph in 1877. Edison strived to connect sound with the picture using his two creations. It would be interesting to see what the film industry would have looked like today if Edison had succeeded in synchronizing the two, almost half a century earlier before it was achieved. Furthermore, even silent movies had sound effects or dialogue produced by people behind the stage.
Another interesting fact is how many sound effects in movies are recorded from real life and manipulated to make them sound unique. Moviegoers rarely notice that some of the effects they hear can be found in real life distorted in thousands of unusual ways. For example, the language of Wookie in Star Wars was recorded from a small bear cub. The way the bears growls were put together to imitate speech completely throws the audience and immerses them into the narrative. It is fascinating to find out how much time the sound director had spent on recording all sounds needed for the movie.
The third interesting fact is the use of ambience in films. It is utilized to set a scenes atmosphere, and it can be very evocative. If used right, ambience is the most critical sound effect, completely immersing the audience into the scene and making it feel as the world on the screen is real. It can also set an emotional tone, reminding the audience members of their own experiences and connecting them to the screens action through it.
Overall, Making Waves provides a compelling description of the evolution of sound in the movie industry. Using the right sounds and music in movies is as important as writing a compelling script and ensuring each scene in a shot with the correct lighting, framing, and composition. Sound, or the absence of it at the right moment, can set the atmosphere and provoke the needed reaction from the audience.
Corporate policies for the use of email to support sound cybersecurity include the following:
Remote Access Policy (RAP)
The mode in which company employees access the internal network of an organization should be guarded by certain rules and regulations
Network Security Policy (NSP)
Computer network access should be restricted by well-outlined rules and guidelines
Email/communication policy
The usage of various communication channels and emails within an organization should be regulated (Yan, Qian, Sharif, & Tipper, 2012)
Internet Access Policy (IAP)
All categories of employees accessing internet connectivity within an organization should follow some well laid down rules and guidelines
Acceptable Use Policy (AUP)
This policy refers to the fair usage of an organization’s cyber network whereby the manager or owner of an organization should apply a set of rules and regulations on how to use the network platform of a firm.
In order to support the email policies outlined above, the IT management team alongside the executive arm of an organization should stress to employees the significance of cybersecurity. Regular workshops congregating all the employees should be organized so that they can be informed about the importance of being secure in cyberspace. Employees should understand the potential risks of cyber attacks (Von Solms & Van Niekerk, 2013). For example, employees should appreciate the fact that the operations of a company can be brought to a halt in the event of a cyber attack. Their individual efficiencies can also be compromised if they do not secure their immediate cyberspace.
Second, all internet users in an organization ought to be taught effective password management practices. The cybersecurity of an organization can be broken down by weak passwords that employees use in their e-mails. Employees should be encouraged to use passwords that cannot be easily compromised. Other password guidelines include using the same or similar passwords in various web pages, sharing passwords, and storage of passwords.
All internet users in an organization should be in a position to detect different types of phishing scams. Employees should be taught how to detect suspicious emails delivered into their inboxes and also avoid opening such emails at random. In particular, using external email architectures (such as Yahoo and Gmail) on a company’s cyberspace should be discouraged. Opening attachments without clearly establishing the sender should be avoided. In addition, confidential company information may be leaked through the phone. Hence, employees should be warned against exposing confidential data to unknown or unwarranted individuals.
Other measures include applying regular updates, protecting sensitive data, and locking computers and other machines used to connect to a company’s network.
A number of supporting guidelines and recommendations to support the mandatory policies, procedures, and standards can be adopted by organizations that desire to secure their cyberspace. For example, all portable media are supposed to be secured (Eastton & Taylor, 2011). Limiting access to portable media like laptops and phones is a major security measure. Before connecting a device to a company’s network, it is highly recommended for such devices to be scanned.
Moreover, stolen or misplaced company’s devices are to be reported to the IT security department as soon as possible. In some cases, attackers can easily gain access to a company’s network through stolen devices (Whitman & Mattord, 2011). It is possible for the IT experts of a firm to wipe the stolen and vulnerable devices before attackers can use them.
Employees and other internet users in an organization may be informed on how to play active roles in the cybersecurity of a firm. For instance, any unusual activity in their emails and connections must be promptly reported to the Information Technology administrator.
When using social media accounts, employees are supposed to apply optimum privacy settings. They should limit access of private information to the outside world except those who are in their accepted contacts (Andress & Winterfeld, 2013).
An organization’s workstations/computers and servers should also be fully installed with patch management applications. In case of any vulnerability, cyber attackers cannot penetrate patched systems.
References
Andress, J., & Winterfeld, S. (2013). Cyber warfare: techniques, tactics and tools for security practitioners. New York: Elsevier.
Eastton, C., & Taylor, J. (2011). Computer Crime, Investigation, and the Law. Boston, MA: Course Technology, Cengage Learning.
Von Solms, R., & Van Niekerk, J. (2013). From information security to cyber security. computers & security, 38, 97-102.
Whitman, M. E., & Mattord, H. (2011). Reading & Cases in Information Security: Law & Ethics. Boston, MA: Course Technology, Cengage Learning.
Yan, Y., Qian, Y., Sharif, H., & Tipper, D. (2012). A survey on cyber security for smart grid communications. IEEE Communications Surveys & Tutorials, 14(4), 998- 1010.
Even large multinational conglomerate corporations have proven to be vulnerable to cyberattacks, exemplified by the information breach suffered by Sony Pictures Entertainment. This attack led to the release of confidential information and a number of unreleased films and caused considerable damage to the company, as well as an uproar in the film industry. However, as noted by the Forbes article “What The Sony Hack Can Teach About Cyber Security”, the companies are unlikely to return to the old ways of managing IT security by closing their information off entirely, due to the public demand for digital openness (Dawson, 2015). This means that the companies have begun to focus their attention on developing other methods of cyber protection.
Developing a company training program to educate employees on cybersecurity policies, procedures, standards, and guidelines to ensure state and federal cyber law compliance is a difficult and a meticulous task, which needs to take into consideration both present and future dangers, the nature of the business itself, and other individual needs and vulnerabilities.
The first step in developing a sound cyber training program requires creating a training program, which would define objectives, within reasonable deadlines. A special group would need to be dedicated to designing and implementing such a program.
The second step would be to establish the conditions of successful cybersecurity programs. To maximize the result the right people need to be selected for the job would. Besides possessing essential IT skills and experience, the personnel involved in the training needs to go through extensive background checks, to ensure no conflicts of interest or even potential malicious intent (Eastton & Taylor, 2011).
The resulting participants will be provided a tool-based and a narrative-based training. Tool-based workshops will focus on mastering the software and hardware used by the company to protect itself against both internal and external attempts of cyberattacks. This training will also provide the staff with information about the types of software that can be used against the company and will be given a chance to acclimate themselves with their functions. This training will include workshops, practices, and role-playing exercises, aimed at reaffirming the learned practices and routines.
The narrative-based training will focus on the theory of cyber dangers and security, and teach them about the tactics used by cyber attackers, the cybersecurity policies, procedures, standards, and guidelines they need to understand and adhere. This would include both internal policies within the company, and external, which include state and federal cyber law. This training will include presentations, lectures, and case studies (Stevens-Adams et al., 2013). Besides dedicated groups, focused on cybersecurity, regular narrative training would need to be provided to regular staff to increase their preparedness. The content of the training would be identical to the narrative training of the dedicated groups and would consist of the same practices.
The cybersecurity training will be conducted upon induction into the company, followed by regular “refresher” training, drills, and assessments. While many sources show companies perform annual assessments, in the rapidly developing cyberspace it would be better to conduct total testing of the staff cybersecurity familiarity at least twice a year and hold briefings once in one-two months, depending on the staff turnover and frequency of attacks (Egan, 2014).
To maximize the effectiveness, the training needs to consist of knowledge, skill, and experience building, and be supported by continuous evaluation of the participants, to ensure their full comprehension and eliminate the risks of vulnerabilities due to gaps in knowledge. The level of employee comprehension and dedication to the cybersecurity policies would need to be regularly monitored, in a way that is ethical, legal, and to which the employees have knowingly consented (Yerby, 2013).
References
Dawson, F. (2015). What The Sony Hack Can Teach About Cyber Security. Forbes.
Easttom, C. & Taylor, J. (2011). Computer crime, investigation, and the law. Boston, Mass.: Course Technology PTR/Cengage Learning.
Egan, G. (2014). What’s Your Frequency of Security Training vs. Frequency of Attack?
Stevens-Adams, S., Carbajal, A., Silva, A., Nauer, K., Anderson, B., Reed, T. & Forsythe, C. (2013). Enhanced Training for Cyber Situational Awareness. Foundations Of Augmented Cognition, 90-99. Web.
Yerby, J. (2013). Legal and ethical issues of employee monitoring. Journal of Applied Knowledge Management, 1(2), 44-55.
Analog to digital converters have become very famous in all parts of the world. Designing your own analog to digital converter needs special equipment, tools and a strong circuit. There are numerous digital to analog converters and processors are available and each of which has its own advantages and disadvantages. Neutral voices follow different paths and different variations due to which human voice faces different pitches and tones.
Variations in human voice pitch and tone affect a lot in the functionality of Analogue to digital converter. There is a strong need to design a circuit by keeping an eye on the variations in input in order to produce effective and accurate output. Usually, mike is used to take input and left or right and sometimes both speakers are used to produce an output. Neutral voices and input in analog to digital converter also change due to environmental factors. Change in human voice due to any reason highly affects the output through an analog to a digital system. An efficient system always has a strong ability to absorb distortion and other noises in input for the production of effective and accurate output.
An excellent analog to the digital processor must pose the ability to produce accurate and sharp output. Generally, analog to the digital processor is used to convert the analog voltage or current in to Digital numbers. Mostly it is an electronic device but there is some non-electric or semi-electric device are also considered as analog to the digital processor. Such devices may include rotary encoders. Gray codes, binary or two’s complement are the popular coding schemes for the digital output. Resolution tells the number of discrete (digital) values a processor can produce for a range of values. Normally values are stored in binary form. It means that unit of the resolution is bit or one can say that resolution is expressed in bits. Formula for the calculation of no of bit in resolution is 2^8= 256 bits.
A good sound system can handle the peak power of an amplifier and also able to deliver it without distortion. By using headroom with an amplifier will give the sound to the satisfaction level without the distortion. Some modern amplifiers have their own headroom so there is no need of external head room. But keep in mind before buying this type of amplifiers output power of the amplifier should be equal to the IEC power of the speakers.
All the natural voices are analog in nature. An analogue to digital processor is used to convert the analog signal (human voice) into the digital signal. Analogue to digital processor works in a very simple manner. It measures the amplitude of an electrical pressure sound wave. As a result measurements in binary bytes are generated. Amplitude is always measured in a sample. Sample rate is the unit of amplitude. Higher sample rate means better quality sound. But it will increase the size of the file. A file with lower sample rate shows that it has poor sound, and small size.
Human ear can hear maximum of 20Hz.
Human audio spectrum = 20 Hz to 20,000Hz (20 KHz)
Highest audio frequency = 20,000Hz
Usually analog to digital process use some circuit to convert the analog data.
“Analog-to-digital (A/D) converters are specially designed to convert analogue input and widely used to transform analog information, like measurements of physical variables, audio signals which includes temperature, shaft rotation and force into a suitable format for digital processing, which usually involve any of following operations: processing by a computer or by logic circuits, storage until ready for further handling, display in numerical or graphical form, and transmission, including arithmetical operations, comparison, sorting, ordering, and code conversion.
If a broad-range analog signal is use for conversion, with a suitable frequency, to a suitable height of two-level bits, or digits, the digital representation of the signal can be transmitted through a noisy medium without relative degradation of the fine structure of the original signal (Asahi 2). Conversion usually composed of quantizing and encoding. Quantizing is usually refers as
“Distributing the analog signal range into a number of discrete quanta and determining to which quantum the input signal belongs”. (A/D Converter)
Encoding means assigning a unique digital code to each quantum and determining the code that corresponds to the input signal (Adrian 346). The most common used system in real world applications is binary system in form of 0 and 1, in binary systems there are 2n quanta(quanta refers to whole number, and the code is a series of n physical bi-valued bits and levels (1 or 0) interlinked with the binary number associated with the signal quantum.
The demonstration shows a traditional three-bit binary presentation of a range input signals, divided into eight quanta. For instance, any signal has the vicinity of 3/8; and on full scale it has vicinity between 5/16 and 7/16 and on both scales it will be coded as 011.
Problem Statement
Objective of this project is to design a sound mixing system that can process two sounds. One of them is analog and other one is digital. System should mix them and after mixing output is send to the speakers. First analog sound will be send to a compressor circuit. Compressor circuit can be controlled manually. Circuit is used to avoid the clipping (Distortion). At the same time analog sound is send to the digital echo adding circuit. Digital echo circuit compresses and adds distortion to the input signal. After sometime output of both the mixers and circuits is send to the main mixer, where it is combined with the digital sound. Source of the digital sound is the I-pod. Digital sound is passed through the tone control module (filter). After passing through the system it is send to the main mixer. Main mixer will be then responsible for combining the two sounds and sending them to the speakers.
Introduction and background
Analog-to digital converter include an analog modulator. Analog modulator is responsible for receiving an analog input voltage and output a pulse train. Pulse train and input voltage has equal proportional value. A digital filter is used to filter the pulse train. After filtration digital filter give the output to the calibration module. Calibration circuit is responsible for controlling the calibration module. It corrects the output and removes the errors. Pre-stored calibration parameters stored in a register are used for this purpose. A control circuit controls the calibration multiplexer. Multiplexer chooses the zero scale reference for the calculation of an offset value and stored it into the register. And select the full scale reference to calculate the scale factor.
The use of analog-to digital converters has been increased in recent years because of advancement of digital signal processing. Increase in the use of digital transmission system is also the reason of this. Usually analog to digital converter consists of a circuit. Circuit receives an analog input signal and output a digital value. This digital value is proportional to the input analog signal. Value can be any thing it can be a word or a digital bit string.
There are different types of analog to digital conversion schemes, some of them are delta-sigma modulation, charge redistribution and frequency converters. Each scheme has its own advantages and disadvantages.
Delta-sigma modulation, this type of analog to digital conversation got popularity in recent years. Analog signal voltage is input into the delta-sigma modulator in delta-sigma modulation. After inputting the analog voltage output is filter to remove the noise. In this type of conversion scheme analog input is converted into the digital pulse string. Pulse string has average amplitude over time proportional to the input. Delta-sigma conversion is popular because its gives high accuracy and wide dynamic range than the earlier delta modulation techniques. Sometime it is also called over sampled converter architecture.
Delta-sigma modulation has two main components or parts: the analog modulator and digital filter. Analog modulator produces low resolution digital output. There are number of noise sources for any analog to digital converter that can mix up with the analog modulator design. In delta-sigma modulation there are chances of noise mixing at the output stage as well as at he input stage. Digital low pass filter is used to filter the noise.
Typically digital signal processor is responsible for the digital filtering of the analog input. Digital data processor at lower rates is well handled by the digital signal processor. The process of output rate conversion is called decimation. In decimation sample rate of a signal is digitally converted to a lower rate (Kevin Nary 3). Usually the decimation process is used to remove the noise form the large amount of the frequency and outputted by a delta-sigma modulator. This process removes the almost 95% of the noise. It can be quiet expensive to implement it in a manufacturing environment. As, it may involve the adjustment of capacitor.
Brief Description
Fig 1 shows the analog to digital conversion. Calibration multiplexer receives an analog input voltage at line 10. Two other inputs can be received by calibration multiplexer 12. Terminal 14 receive full-scale Reference and terminal 16 receive zero-scale reference. Analog modulator 18 receives the out put of calibration multiplexer as input.
Analog input is converted into the pulse string by the analog modulator 18.This pulse string have an average amplitude over time proportional. Usually delta-sigma modulator realizes this. But other modulators cal also realizes or provides this type of relationship. For instance, a voltage-to-frequency converter (V/f) converter. Voltage-to-frequency converter can also convert the analog input signal into a frequency. This frequency can provide the digital output by sampling.
There are chances of noise association such as quantization noise because of the predetermined sample rate of analog modulator 18.
A digital filter is used to remove the noise and to provide a digital output. Digital output is provided on a digital bus 22.
Calibration module 24 receives the output of the digital filter 20 as input. After receiving the input calibration module 24 provides the output on an output data bus. Calibration module 24 can provide a dc offset correction in simplest mode. High linearity is exhibited by the analog modulator 18.But calibrator module 24 can be used for the nonlinearities in the output signal. Calibration control circuit 28 controls the calibration module 24 and interfaced through a bus 29. Calibration control circuit 28 stores calibration parameters in the memory register. It also controls the calibration multiplexer 12 with the help of a control bus 32.
In simplest mode calibration control circuit 28 access the calibration parameter stored in the memory register 30 and also controls the calibration multiplexer 12, so that it can provide a correction. External calibration signal CAL selects the calibration mode. And inputs to the calibration control circuit 28 from the exterior of analog to digital converter through a line 34. Analog to digital converter generate the calibration parameters automatically for storage in register 30 after receiving the signal.
Figure 2a shows the relationship between the digital output value and analog input value. Dotted curve at line 36 shows the relationship between them. Solid lien 38 represents the uncorrected value. It clearly indicates the dc offset and gain variation. That results in a different slope. Calibration control circuit 28 controls the calibration module 24 for the measurement of output value of digital filter 20. Calibration control circuit also calculates the difference between the measured value and the expected value. This is known as offset. Memory registers 30 is used to store the offset. Fig 2b shows the correction of this offset in the solid curve 40. Dotted line 38 shows the actual measured value. After making the offset correction it sis stored in the memory register 30.
After the settlement of the digital filter 20, output as the offset value is stored in the memory register. Due to this calibration module 24 and the calibration control circuit 28 can check that whether there is a difference between the expected output value on the ideal curve 36, and the actual measured value of the on the curve 40. Difference tells that there is an error. Approximation technique is used to determine gain scaling factor. Then gain factor on a scale is stored in the memory register 30. Calibration control circuit 28 controls the calibration module 24 to correct the output of the digital filter 20. Curve 42 represents the resulting offset and gain correct output. All the curves in the figs 2a-2c illustrate the uni-polar operation.
Fig 3 gives more detailed over view of analog to digital converter. As described above, in delta-sigma modulation analog modulator input an analog voltage and change or converts it in to the digital output. Seven bits digital word output on a digital bus 46 by the FIR 44 is of great significance. Decimation function is provided by the FIR 44.sample rate of analog modulator 18 is 16 kHz but it is decimated to 4 kHz rate by the FIR 44. Response filter (IIR) 48 takes seven bit input from the bus 46 and provide 28 bit output on bus 50. Major portion of filtering is provided by the IIR filter 48. The resolution of filters can be improved or increased on bus 26.
Calibration control logic circuit 52 and calibration logic circuit 28 receive 28 bit as input from the bus 50. Calibration control logic circuit 28 provides input to calibration logic circuit 52 on bus 54. Time required for analog to digital conversion is also determined by the calibrating control logic circuit 28. FIR 44 and IIR 48 both can input voltage but either on terminal 14(full scaled reference voltage) or on the terminal 16 (zero scaled reference voltage).
In Fig 3 it looks like that terminal 14 is connected to the V ref and terminal 16 is connected to the ground. 1024 operational cycles are must before the settlement of the output. 1024 cycles are must to calculate the offset. And to calculate scale factor 1024 more cycles are required. Another 1024 cycles are needed to return to normal operation. Input voltage V1n is inputted to the terminal 10. There is an important point to note, that all the operation is automatic in self-calibration mode. Self calibration sequence is initiated by the CAL signal on line 34.It calculates and store the dc offset value. It also calculates the scale factor. Three cycles are necessary to set the counter 56.
Expected Complete System Diagram.
Circuit Diagram
Input mixers
A detail of simple 3 line inputs and 3 mike inputs is given below:
It can be used with the Analog to digital processor. The following figure shows that the mixer circuit has thee inputs and three mikes inputs. 200-1000R dynamic phones can be used as with it. They are best suitable microphones with this input. ECM or condenser mike can also be used. But it’s the second option and it must have bias via series of resistors. There will be a slight loss. But it’s not a big problem because it will be with the every mixer circuit.
A gain of 2 or 6db can be summed to the final amplifier handle this problem. Keep in mind that the input level should must be the approximately 200 m V RMS. The mike inputs are amplified to the 40 db. In this way total gain with the mixer can be 46db. The mike inputs are designed in such a way that microphones give 2mV RMS at a combine or a single meter. Majority of the microphones meets this standard.
Op-amp is not the only solution for this type if circuit. MOS type op-amps or bipolar, FET can be a good substitute. Two batteries of 9 volts can be used but if want to use for longer periods then power supply is recommended. The power supply should be dual negative and positive supply. As shown in the above diagram.
Another input mixer can be also be used. This is discussed below.
It’s a simple input mixer. It contains two transistor based preamplifiers.
One can be used for the higher gain. As shown in the diagram as “Mike in”. Norma regular microphone works best with it. The other “audio in 100n” can be used to control the input form any other source like CD player or tape player. Keep in minds that don’t place the circuit to much close to the main transformer. Modification can be made to it to control it better. But it can also be controlled manually, with out any modification.
In fig 4, The FIR filter 44, is highly depend on FIR digital signal processing which is linked with an attached FIR controller 60 for effective results through bus 60. The whole circuit is interconnected in order to process analog input to digital sound. The constants of filter 44 are linked with the coefficient of ROM 64. The constants of filter 44 are in the FIR constants Rom 64 which is interlined with a FIR controller 60 through a data bus 66. It is always preferable in order to create an efficient and time saving system to use buses and FIR controllers so that voice can be changed within a seconds. As much the system is strong the output would be strong and effective. Use of unnecessary buses and controllers usually make system messy and inefficient. In best incarnation the ROM 64 is linked with a prologic array.
Superb calculations are required in order to produce effective and strong mixing system. For the various types of inputs it’s necessary to use strong and efficient buses and logic array in order to make sound system effective. Logic array has of great significance and provide a suitable pathway for setting up a strong circuit. The FIR DSP is significantly important and an essential part in designing digital to analog processor.
THE FIR DSP is an Arithmetic logic unit which takes input for multiplexing in order to perform the calculations important to realize the functionality of the system. Basically, digital filters are composed of series of arithmetic functions i.e. multiplication, addition and subtraction steps which are necessary to execute in a sequenced order. The serial data stream from moderator 18 is executed through FIR DSP 58 for the storage of coefficients in the ROM 64.
This series provides an output on bus 46; bus 46 is comprised of seven bits which are used to save data for further processing. Storage on bus 46 at a rate of 4KHZ as compared to 16KHZ output by the modulator 18. The FIR DSP 58 provides dissemination of the output modular 18 on 16 KHZ rate to a 4KHZ rate. The IIR filter and circuit 52 are made with the aid of IIR/CAL digital signal processor (DSP) 68.
The IIR/CAL 68 is exclusively designed in order to control the operation of DSP 86. The designed DSP 68 is linked with the controller 70 with the aid of two dimensional data. The control bus 72 and the controller 72 is linked with an IIR coefficient ROM 74 with the aid of bus 76 and with the combination of bus RAM through two dimensional data bus 80. The IIR CAL DSP 86 has of great significance in overall circuit. It is specially designed keeping an eye on change inputs.
The IIR/CAL DSP 86 is a strong arithmetic logic unit which owns input for multiplexing to provide a sequence of multiplication and other arithmetic operations according to IIR algorithm. Once the IIR function is complete, results are used for producing different outputs. Once the filter unction of IIR the results are usually used in order to perform the calibration and correction function with the aid of same arithmetic and logical unit. The single arithmetic unit provides the filter function the calibration function and the output correction function. While processing, some units and data are stored in ROM 64 for further processing.
According to IIR algorithm it is an important factor to store coefficient in Rom 64 and utilized them in later stages. The output of the IIR /Cal 68 is an input of parallel to serial converter 80 that which is use to convert 28 bit output in to 16 bit serial data string. As it is states above that calibration and output correction function are operated at twenty eight bit resolution and as it has been observed that output is on 16 bit resolution.
A flow chart shows the calculation for self calibration operation. According to which calibration starts at block 90 and travels to decision block 92. If “Y’ is not present then input travel back to the input of decision block 92 through “N” path. When Cal signal is present, usually a zero scale value is selected in a function block 94. After selection of zero scale value, the program offers a delay of 1024 cycles, and then the program flows to the function block 98 to calculate and store the offset value.
In following Fig a flow chart for the output correction process is shown after the storage of the parameters in the calibration RAM 78. The input usually start at block 116 and flows back to block 118 in order to fetch the offset value from the calibration of RAM 78. If system is bipolar then the input usually flows along the “Y” path from the 121 de4cision block. In the bi-polar mode the sequence of input and output is slight differ from polar mode. Sequence of input has of great importance in overall circuit. A strong circuit helps in producing effective and strong results. Both bi polar and polar mode processing is different in sound mixing system.
The analog to digital converter needs strong interfacing and an efficient bus sequences so that the voice can travel in the right direction and chooses straight path. The analog to digital processor of claim one receives input comprises of a multiplexer i.e. having one input and multiple outputs. A self calibrating analog to digital converter is usually comprised of a modulator for receiving analog input and outputting a digital signal proportional to said analog input signal.
Audio input mixer is easy to design it usually needs two transistor based preamplifiers. The one is of higher gain and second one is usually low. Sound input mixer usually works with a regular dynamic microphone. The second transistor is sued to control audio input from different sources like tape or recorder or CD player etc. Audio mixing needs special attention in order to produce effective and efficient results.
Audio mixing is a procedure by which a multitude recorded sounds are usually combined into one or more channels. Multi-track recording can also do by the aid of following system. Generally, different types of equipments are used in mixing system. Basically there are three process involved in sound mixing1) Mixing 2) Routing 3) processing. All three sub processes are important and play pivotal role qt mixing console. Following equipments are generally used in audio mixing: Mixers Outboard gear and plug-in, Processors, Faders, Compressors and Equalizers. Live audio mixing is also another aspect of audio mixing.
Live audio is a process in which a multitude of sound sources are combined through one or more channels in a live performance. The whole process of live mixing is done via sound engineering or recording engineering. The best example of live audio mixing is a live concert in which two mixers are involved one is fixed in the audience to mix front of house speakers and other is usually located at the side of the stage. Mixing of speakers is directly positioned in front of the performers in order to make sure that they can hear one another. A separate mixer of broadcast can also be fixed.
Tone Control Module
A simple but powerful tone module control is shown above. It is useful for many applications. It’s up to the owner that he uses this as a stand alone module controller or adds it to the amplifiers. Also it can be used with any analog to digital processor. One interesting feature is that it can also be used to build new and exciting instruments. One IC shows that dual power supply is not must for it. 9 to 15 volts are enough to use this. But remember that bass will be weak in case of 9 volts usage.
Table shows that hardware information.
Parts.
Quantity
C1,C3,C5,C7,C15, C16
6
C2,C6
2
C4
1
C8,C0
2
C9
1
C11,C12,C13,C14
4
R1,R4
2
R2,R5
2
R3,R6
2
R7
1
U1
1
S1
1
R8,R9,R10,R11
4
R10 is used to control the balance. R9 is used to control the bass.R8 is used to control the treble.
J3 and J4 are the right outputs while J1 and J2 are the left inputs. A voltage divider is must for a stronger signal.
Here is another tone control modular discussed below.
A high quality performance, Portable mixer that uses 9 volt battery can be designed easily. Three main modules can be assembled to it according to the needs.
Input Amplifier Module
Tone control Module
Main mixer Amplifier Module.
Tone control module will only be discussed here.
Tone Control Module
It’s a three band Control circuit. Three bands will be (bass, treble and middle). Unity gain is provided in response of the flat frequency. It can be inserted after the main amplifiers but also can be used independent with any analog to digital processor.
Fig: 4 Circuit Diagram of Tone Control modules.
It is a simple circuit designed to control the three bands. When the control module in the center then the voltage gain will be 1.
Parts
P1, P2_________100K Linear Potentiometers
P3____________470K Linear Potentiometer
R1, R2, R3_______12K 1/4W Resistors
R4, R5___________3K9 1/4W Resistors
R6, R7___________1K8 1/4W Resistors
R8, R9__________22K 1/4W Resistors
R10___________560R 1/4W Resistor
R11___________100K 1/4W Resistor
R12___________220R 1/4W Resistor
C1______________1µF 63V Polyester Capacitor
C2_____________47nF 63V Polyester Capacitor
C3, C5___________4n7 63V Polyester Capacitors
C4_____________22nF 63V Polyester Capacitor
C6, C8_________100µF 25V Electrolytic Capacitors
C7______________4µ7 63V Electrolytic Capacitor
IC1___________TL061 Low current BIFET Op-Amp
Note
Frequency response = 20 Hz- 20Khz
Tone control frequency = ±15dB @ 30Hz; ±19dB @ 1KHz; ±16dB @ 10KHz
Distortion at 2 v RMS = <0.012% @ 1 KHz; <0.03% @ 10 KHz.
Maximum Distorted output = 2.5V RMS.
Microphone Preamplifier
It’s a simple microphone pre amplifier. It doesn’t use any special protection circuit. It’s one transistor circuit. Its frequency response is 20 Hz to 20 KHz tat makes is little bit noisy. It uses the commonly available components. A line level input or microphone or computer sound card can be used. Less than 10mA current is required for it. It will cost you a total of $ 10. So it’s not a huge amount. It’s cheaper than any market pre microphone amplifier.
It can be used between the amplifiers or analog to digital processor and microphone. It can take both the dynamic and electric microphone. It is kept simple so that every can make it easily. This circuit is simple it uses one transistor amplifier and amplifies about 30-40 db. LED D1 shows that circuit is operate able. Drop in voltage is caused by the LED (around 1.8V for RED led). Resistor R4 and capacitor C5 are used to filter the noise. It tries its best to reduce the noise to maximum possible level. DC bias is blocked by the capacitor C1, C2 and c3. Because it’s a very simple and a cheap circuit. So the distortion performance is not so good. There will be distortion of 2-3%.
Attenuation is another problem. Attenuation is caused by the sound blaster. Microphone preamplifier circuit generates bass frequency. List of components is given below.
R1 4.7 kohm
R2 220 kohm
R3 2.2 kohm
R4 120 ohm
C1..C4 10 uF 16V electrolytic
C5 100 uF 16V electrolytic
D1 Red LED
Q1 BC547B
SW1 on/off switch
May be some components are not available in the market easily. Then other suggested components can be used. For example if BC547 is not available then, 2N222. It also works well. Value of R2 can be changed between the 100 kohm and 470 kohm.
R1, c1, and C2 can also be left. One can leave them to make it more simple circuit.
Versatile Compressor
It will be versatile compressor circuit that take the analog data as input and compress it to avoid the clipping. It protects the analog data from distortion. This circuit is controlled manually. One can set the compression according to its needs. At the same time there is another circuit that is responsible for the echo.
Digital Echo
IC 1 is worked as clock generator, which runs at the speed 4 tie greater than the sampling frequency. Frequency can be adjusted by the VR1. This unusual configuration generates output with equal mark space ratio. IC7 (A-D) converter need a negative bias on pin 5. As the voltage Requirement is very low. Sop this negative voltage can be obtained by rectifying the clock signal from IC1; it gives approximately -4 voltages.
There is another IC named IC2. It produces the four pulses. Three of them are inverted by the gate in IC3. A-D processor IC7 starts a conversion when Q1 goes high. After the data in Q2 is high, all the data of RAM sent to the D-A converter IC8. It produces the required voltage. Data is stored in the Ram when Q3 is high (Mark 1). Address counters IC4 and IC5 got increment because of the Q4. Whenever Q4 goes high it causes increment in the address counters IC4 and IC5.
Amount of the Ram chip to be used is set by the SW2. Sw2 detect the address counters before they are reset. If there is continuously variable delay pot on the 555 clock circuit it’s the ideal condition. But it’s not possible to achieve the required range. Analog data is input in the versatile compressor circuit that compresses it.
At the same time data is sent in the echo circuit that adds the digital echo.
After if the analog data is send to the main mixer where it is mixed with the digital data and after mixing sends it to the speakers.
Threshold
Analog processor is used to adjust or manage the threshold. Automatic gain control topology is used for the compression. Threshold adjustment permits the operator to select that level at which compression starts.
Baxandall control
Simple resonant network is usually used for the active lc design. Band pass filter function is created by the LCR resonant circuit. Fro this purpose a famous topology is used that is based on the Peter Baxandall’s famous negative feedback tone control as shown in the diagram below
LC equalizer based on Baxandall negative feedback tone control circuit
Buffer
Buffer may be used as a device for short radio signals in the internal control unit.
Dynamic range
Problem of dynamic range is solved by using the multi-band compression schemes.
Minimum voltage range used in quantization process is determined by the bit resolution. And
Can be represented digitally.
Ratio
Compression ratio is also managed by the analog processor. Ratio of compression can be described as the ratio of total gain reduction of the input signal to output signal. Input output level is used to measure the amount of the ratio numerically.
Attack
Attack time is adjustable, it allows the users to adjust the short term loudness according to their requirement or need.
Digital delay echo
AD converter is used to convert the input signal into the digital signals. After it supplied to a feed back loop that have a digital delay circuit. In this ways echo is produced. There is no special or separate device for this purpose.
Amplifier mixers what they do speakers
Amplifier takes the weak input signal as input mixes it and boosts it. So that it can have enough power to drive a speaker.
Performance Equipment
There are meters that show the performance of the system. There are numerous equipment involve din overall circuit. The overall system needs strong and effective interface in order to process analog sound (human sound) into digital sound. As it has been observed in the overall system, numerous equipments are linked together. Buses, RAM, ROM and controllers play pivotal role in overall system. Performance is highly based on their setup. Overall, system setup is exclusively designed keeping an eye on various factors like variation in sound input. There are a number of variations involved in analogue input.
Pitch and tone highly affects on the output. Variations in pitch and sound input have a great impact on output. This is the only reason why analogue sound is difficult to deal with. Tone and variations in pitch usually causes problems while passing through circuit. Buses and controllers required a smooth flow in order to produce effective output from a speaker. Overall, performance of equipments is highly based on complete setup of the circuit. When processing sound homogenous mixing of components is extremely important. There are certain conditions on which system works extremely fine and in some cases equipments and overall circuit doesn’t work smoothly. There are some limitations like, smooth flow of input, clear and up to the mark input, buses and controlled should be interfaced accurately.
Testing procedure
Testing procedure based on the Histogram method. It uses triangular waves of small amplitude. It has several advantages as compared to the traditional static test.
A dynamic test can be performed by the use of Gaussian noise this can determine the nonlinearities and compensating.
Analog to digital processor can be best tested by combining a histogram-based approach and spectral analysis. It completely checks the analog to digital processor. Testing procedure normally checks for the distortion.
Expected problems
A problem can arise because of its nonlinear nature and feedback, because of limit cycle oscillation. It will produce a non desirable tone. This tone is not bearable in this type of application. There are many other problems can be found in analog to digital processor, some of them are given below.
Aperture Delay
DC Common-Mode Error: – It is expressed in the LSBs. It occurs due to the change in the analog input voltages. It also affects the output.
Differential Gain Error:- It shows the difference between the output amplitude in percentage.
Differential Phase Error:- Difference in the output phase of small signal sine wave is called differential phase error.
Full Scale Error:- It is measured by the Vmax+1.5LSB-Vref.
Gain Error:- To determine the gain error its formula is Vref-1.5LSB
Missing Codes
Offset Error:- It’s the difference between the actual transition and ideal LSB transition.
Output Delay: – time delay
Pipeline Delay
Tolerance Analysis
Tolerance analysis is the key part of all systems. It indicates and helps in determining the strength of a system. It also helps in evaluation of the system. The following system gives complete control on devices. All devices can be handled manually. All setup is completing reprogrammable, it is a stand alone system, and it’s a complete portable unit. It has strong control on overall flow. Microprocessor and A/D converter has fault tolerance capacity and have ability to handle strong and low pitch control.
Testing phases gives complete evaluation of the project. IT is always recommended to make test cases in order to judge the fault tolerance and functionality of the overall system. A key part of the entire system is to compare the incoming data stream with the stored data stream. A tolerance must be allowed in order to match words. But the care should be taken that word sound similar will also work on the particular system. Fault tolerance of the following system gives a complete idea of system evaluation. The research done with the use of different techniques will conclude how much tolerance is allowed in the particular system.
Analog footage in opposition to digital soundtracks compare the two forms in which resonance is documented and stockpiled. Authentic jingle waves consist of continuous discrepancies in air anxieties. To illustrate these indicators can be documented in, digital or analog layouts. Analog documentation is one where the inventive sound indicator is adapted onto a subsequent corporeal channel or substrate such as the groove of a turntable compact disk or an iron oxide surface of a magnetic ribbon, (Rumsey & Watkinson 1995, Dunn 2003:34). A substantial eminence in the standard is straightforwardly associated or resembles, to the tangible attributes of the inventive soundtrack; thus the amplitude phase and so forth. Digital documentation is produced by converting the tangible attributes of the inventive sound into a sequence of integers, which can then be stockpiled and replayed for facsimile. The truthfulness of the translation process is dependent on the case tempo and the case profundity, thus, the quantity of in rank comprised in each sample, this can be illustrated as the optimum integer extent each mock-up significance. Nonetheless, unlike analog documentation that is importantly dependent on the long term durability of the fidelity of waveforms documented on the standard, the tangible standard stockpiling digital samples is fundamentally immaterial in replaying of the encoded in rank so long as the original information of numerals can be improved, Rathmell, J. et al. (1997:832-840).
Major Discrepancies
This domain has been under discussion as to whether analog acoustic is superlative to digital auditory or subordinate versa. The subject is extremely reliant on the distinction of the structures under analysis as well as other variables that are not significantly associated with resonance excellence. Squabbles for analog structures incorporate the absence of fundamental error apparatus that are present in digital acoustic structures, this include aliasing, quantization racket in addition to presumed inadequacies in self-motivated assortment.
Supporters of digital tip to the elevated echelons of presentation probable with digital acoustic includes; excellent linearity in the perceptible posse as well as diminutive echelons of noise and distortion, Rathmell, J. et al. (1997:832-840). Precise elevated eminence sound reproduction is probable with analog as well as digital configurations. The most off-putting variable of analog expertise is the analog sensitivity media to corporeal dilapidation. The most significant benefit that digital structures have is a homogenous source fidelity, cost effective media replication expenses, direct functionality of the digital indicator in modern day popular portable storage in addition to replay apparatus. Analog documentation by comparison needs a comparatively bulky, elevated-quality replay functionality to get hold of the indicator from the medium as precisely as digital, (Ramsey & Watkinson 1995, Dunn 2003:34).
Analogue apparatus flow might bring about distortions such as knockout, flurry, and ribbon whisper or in the event that the medium becomes worn, facade racket. Most of these deformations can be tackled by employing moment foundation modification, as is done in VHS tapes, filtering or elevated-eminence constituents. Instance-instability in PCM digital structures (jitter) may be audible one some indicators, particularly sinusoids (Rumsey & Watkinson 1995, Dunn 2003:34).
As of 2008, all audiophile as well as clients ranking digital structures now predetermine the chronometer into the hinted statistics itself.
In the initial developmental phases of the Compacted disc, designers realized that the aptness of curved of bits was significant to replay dependability. A scrape the dimension of a human hair (100 micrometers) could easily corrupt dozen bits, the resulted in at best a pop, and far worse, a loss of synchronization of the chronometer as well as statistical, giving a long fragment of racket until resynchronized, Rathmell, J. et al. (1997:832-840).
This has been attended to by encoding the digital stream with a multi-tiered error-correcting coding scheme that minimizes CD ability by approximately twenty percent, although it creates it tolerant to myriad of surface imperfections across the disk minus loss of indicator. Quintessentially, mishap rectification can be viewed as employing the arithmetically encoded backup replicas of the statistics that was tainted, Rathmell, J. et al. (1997:832-840).
Miscalculation rectification enhances digital layouts to accept moderately a smidgen more media corrosion than analog configurations. This does not underscore the fact that scantily produced digital media are immune to statistical trouncing. Laser putrefaction was most troublesome to the Laser compact disc layout, and this happens as a result of deficient disc processing.
Intermittently there can be intricacies associated to the functionality of client rewritable CDs. This could be as a result of meager value CD recorder ROM or stumpy-value discs.
Disparate analog replication, digital replicas are normally accurate carbon copies, which can be replicated indefinitely without degradation, except DRM curbed are well-designed or mastering boo-boos transpire.
Digital structures have the ability for the same standard to be employed with arbitrarily elevated or low eminence encoding lines of attack and number of conduits or other content, unlike instinctively pre-rigid tempo as well as conduits of virtually all analog configurations.
Non acoustic benefits of myriad digital structures pragmatically exist. Random access makes editing much easier with respect to their disk as well as memory centered nature, thus allowing the listener towering versatility when opting for tracks. Mainly digital structures accept non-audio statistics to be encoded into the digital stream, this include information regarding the musicians, pathway titles, and so forth, Rathmell, J. et al. (1997:832-840).
Noise and distortion
During the recording process, storing as well as replaying of the inventive sound wave correspondence, the occurrence of digital degradation is inevitable. This deprivation is in the form of linear as well as non-linear inaccuracies (discrepancies to the amplitude or segment rejoinder within a particular bypass band). Racket is not related in time to the inventive signal content, whereas distortion is in one way or another associated in time to the original pointer substance.
Digital Essentials
To begin with, the input of the analog indicator is needed; this indicator emerges directly from a microphone pre-amp, although any analog audio indicator can be altered. The strength of the signal is then determined at standard intermissions, case by the analog-to-digital renovator.
At every casing tip the indicator ought to be assigned a specific intensity from a cluster of assortment of principle quantization. The inventive jingle signal can be illustrated by employing only integers as digital in rank.
The numerals of the illustration are an analog of instance, and the intensity of the samples is an analog of demands at the microphone. In the event that the inventive wave is altered into binary numerals successive augmentation of clatter and buckle can be disallowed at every phase of dispensation. Inaccuracy rectification set of laws fundamental when transferring digital auditory over deafening conduits, assists in abolishing tad faults. Digital in rank is changed into a nonstop, analog signal by a digital-to-analog adaptor, especially when replaying a digital soundtrack. The electronic pointer is then amplified and changed into acoustic signal by the amplifier. Unavoidable racket inherent within acoustic waves include, automatic, electrical as well as thermal noise echelons in the recording as well as replay sequence (automatic transducers (microphones, amps), loudspeakers, recording equipment, mastering procedure, facsimile apparatus, etc.)
Even if the aural wave is at the similar phase, altered into a digital shape will impact how much racket is augmented. The tangible procedure of digital adaptation will constantly augment some racket, even if it is stumpy in strength. The quantity of racket that a filament of aural paraphernalia augments to the inventive wave can be measured. Numerically, this can be illustrated by means of the sign to noise relation (SNR). Occasionally the optimum possible dynamic range of the configuration is cited as a substitute. The amount of crumbs inherent in digital structures where a wave is allowed to have on quantization normally has a bearing on the echelons of racket as well as deformation enlarged to that gesture.
A digital structure that has a 16-bit composition of Red Book audio CD has 2 to the power of sixteen, equivalent to sixty five thousand five hundred and thirty six. (216= 65, 536) probable signals amplitudes methodically accepting for an SNR of 98 db (Sony Europe 2001) as well as the dynamic range of 96 dB
In bid to meeting the hypothetical functionalities of a 16 bit structure, for a 0.5 V crest to climax input contour gesture, a PCM (tempo parameters intonation) quantizer would require a comparable lowly input sensitivity of simply 7.629 microvolt. This is an equivalent of 15.3 ppm sensitivity for a recorder that is analogue by part of the whole recording structure and medium. Digital arrangements of replication are dependent on the analog-to-digital and vice versa alteration phase, as such this does not depend on the eminence of the soundtrack standard.
Characteristically, anything beneath 14 bits can bring about minimized reverberation eminence, which has 80dB of SNR that is termed as an informal ‘minimum for Hi-Fi audio. Nevertheless, it is uncommon to find digital media specified for less than 14 bits, except for older 12 bit PCM camcorder audio (or DAT in long-play, 32 kHz mode) as well as the output from grown-up or inferior-outlay computer software, sound cards/circuitry, and consoles and games (typically 8 bit as a minimum and standard, though trick sample output apparatus for collectively non-PCM hardware gave SNR functionality more rapidly to that of an superlative 6 or 4 bit PCM digital adapter and the output from older or lower-cost computer software, sound cards/circuitry, consoles and games (typically 8 bit as a minimum and standard, though trick sample output methods for generally non-PCM hardware gave SNR performances closer to that of an ideal “6” or “4” bit PCM digital converter).Each supplementary quantization bit methodically adds a 6 dB in probable dynamic range, e.g. 24 x 6 = 144 dB for 24 bit quantization, 126 dB for 21- bit as well as 120 db for 20-bit, Rathmell, J. et al. (1997:832-840).
Analog systems
Punter analog videotape ribbons may possibly have a vibrant assortment of 60 to 70 dB. Analog FM broadcasts rarely have dynamic dimensions beyond 50 dB. The dynamic dimension of a straight-engrave album record might surpass 70dB. Analog igloo master tapes employing Dolby- A racket diminution can have a vibrant range of approximately 80 dB.
Thunder
‘Grumble ‘is a structure of noise attributes of poor or worn gramophones. Owing to imperfections in the bearings of gramophones, the platter happens to have an insignificant quantity of monitor unlike the require cycle. Thus, alongside its cycle, the phonograph also moves patchy as well as plane-to-plane faintly. This supplemental activity is augmented to the required wave as racket normally of very low concurrencies, creates a rumbling’ sound in the event of quiet passages. Incredibly low priced gramophones sometimes employed ball bearings that are most probably to create audible rations of grumble. Additional classy gramophones happen to employ massive sleeve bearings that are much less likely to generate offensive quantities of rumble. Increased gramophones gathering also tends to lead to minimized grumble. Proficient turntable might have rumble at least 60 db below the specified output echelon from the raise up, (Driscoll 1980:79-82)
Consciousness and Shimmer
Wow as well as fluttering are the outcomes of imperfections in the mechanical performance of analog mechanisms. Wow and tremble is most conspicuous on indicators which represent untainted tendencies. As an illustration, 0.22 percent (rms) wow may be detectable by listeners with piano music, although this develops to 0.56 percent with jazz harmony. For LP records, the eminency of the gramophone will have a large impact on the echelons of knockout as well as compliment. A superior gramophone will have wow and flutter values of less than 0.05 percent, which is the speed discrepancy in assessment to the ideal significance.
Occurrence retort
The rate of recurrence reaction of aural CD is sufficiently wide to cover the entire audible range, which approximately protracts from 20 degree Hz to 20 degree kilo hertz. Analog aural is limited in its probable frequency response, although the limitations of the fastidious analog shape will offer a restriction. For digital structures, the optimum aural frequency response is hardcode by the casing regularity. The option of mock-up tempo employed in a digital structure is centered on the Nyquist-Shannon variety theorem. This stipulates that a sampled wave can be duplicated exactly as long as it is sampled at a regularity superior than two times the bandwidth of the wave.
Consequently a sampling tempo of 40 degree kHz would be adequate to capture all the in rank contained in a wave having frequency bandwidth up to twenty percent kilohertz. The intricacy emerges in removing all the wave content beyond twenty percent kilo hertz’s as well as unless this is done aliasing of this elevated tempo might happen. This is normally where these elevated, inaudible tempos alias to frequencies that are in the perceptible assortment to preclude aliasing, it is not necessary to design a brick-wall anti-aliasing filter that is a filter which perfectly removes all tempos content beyond a certain range. As a surrogate, a sample tempo is normally chosen; this is above the methodical obligation. It is referred to over sampling, and allows a less relentless challenging-aliasing sieve that is employed. Soaring eminence open-reel ribbon frequency response that protracts from ten percent hertz to beyond twenty percent kilohertz.
The linearity of the response might be indicated by presenting in sequence on the echelon of the rejoinder relative to allusion regularity. For instance, a structure compound might have a rejoinder denoted as twenty percent hertz to twenty percent kilohertz +/- 3 dB relative to 1 percent kilohertz. Certain analogue proliferators specify tempo rejoinders that reach twenty percent kilohertz, although these measurements might have been made at low signal echelons, (Driscoll 1980:79:82). Soaring quality metal particle cassettes might have a response that stretches up to fourteen kilohertz at full (0.dB) sound tracking dimension, (Stark 1989:625).
The regularity tempo for a traditional LP player may be thirty degree hertz to twenty degree kilo hertz +/- 3 dB. Disparate to the aural CD, vinyl records do not require a cut-off in response beyond twenty percent degree kilohertz. Stumpy regularity response of vinyl records is restricted by grumble clatter. The Compact Disc in comparison presents a tempo rejoinder of twenty degree hertz-20 kHz ± 0.5 dB, with a superlative dynamic range over the complete perceptible tempo gamut. With album verifications, there exists various trounces in fidelity on every playing of the disc. This is due to the show off of the stylus in contact with the record facade. First rate eminence stylus, matched with a correctly sketched pick-up arm, ought to cause minimal facade wear.
In the event that the compact disc is played, corporeal contact does not exist; as such statistical data is read optically by employing the laser sunbeam. As such no such media deterioration is evident, and the Compact Discs will, with proper care, sound the same every interlude it is played.
Analog benefits
It can be disputed that analog layout maintain certain inherent benefits over digital arrangements. The importance of these benefits is anchored on the quality of particular digital as well as analog equipment.
The benefits of analog structures are captioned beneath:
nonexistence of aliasing deformation
deficiency of quantization racket
performance in surplus conditions
Aliasing
Dissimilar to digital aural structures, analog structures do not need filters for band restricting. The riddles function to preclude aliasing deformation in digital apparatus. Antique digital structures might have suffered from various wave distortions associated to the application of analog anti-aliasing filters, thus, time dispersion, non-linear degradations, temperature dependence of filters and so forth. (Hawksford 1991:8).
Jitter
One facet that might preclude the presentation of pragmatic digital structures from meeting their methodological performance is jitter. Jitter is the acronym assigned to the phenomenon of the variations in putting, of the discrete values found in a stream of bits that constitute to a digital wave. This might emerge owing to the timing inexactness of the digital chronometer. Idyllically, a digital clock ought to produce a timing throb at precisely standard intermissions. Induced statistics also represent diverse sources of jitter within digital electronic circuits. At this point, one part of the digital strip impacts on succeeding phase as it flows through the structure and power supply encouraged jitter, Dunn (2003: 20-22). Precision of a digital structure is embedded on the sampled ideals referred to as quantized principles that exist in the amplitude realm although it is anchored on the timing promptness of detached standards that subsist in the sequential realm. The reliance on precision of discrete values in the sequential state is inherent to digital recording and playback and does not have analog equivalent. Intermittent jitter brings about adapted racket and can be viewed of as being the equivalent of analog flicker. Indiscriminate jitter changes the racket base of the digital structure. The sensitivity of the converter to jitter is dependant upon the blueprint of the adaptor. It has been shown that an indiscriminate jitter of 5ns might be importantly for 16 bit digital structures. Since a more detailed description of jitter conjecture, refer to Dunn (2003: 20-22). Jingle eminence in digital aural structures cam be demeaned. In 1998, Benjamin and Gannon reviewed the audibility of jitter employing listening tests. They determined that the lowest echelons of jitter to be audible was around 10ns (rms) thus on a 17 kilohertz sine signal test wave. With aural, no listener established jitter audible at echelons less than 20ns. Ashihara et al (2005:70-7) manuscript attempted to determine the detection thresholds for indiscriminate jitter in melody waves. These systems constitutes of ABX listening tests.
Hitherto genuine jitter in clientele products appear to be too miniaturized to be identified at least from facsimile of aural waves. What has not been clear is whether the detection of threshold acquired in the modern study would actually underscore the restrictions of auditory resolution or it would be curtailed by resolution of apparatus. Deformation owing to very diminutive jitter might be stumpy than distortions owing to non-linear attributes of amplifiers, Dunn (2003: 20-22). Ashihara and Kiryu analyzed linearity of amplifiers in addition to headphones. Basing on their analysis, headphones appear to be more preferable to produce proficient aural force at the ear drums with diminutive deformations than amplifiers. In the event that jitter rations are very high, as in very low cost Compact Disc players (2ns), the outcome is somehow analogous to knockout and flicker, the well known anomaly that affected typically compact discs as well as creating by non perfectly constant tempo of the tape, the impact is analogous, although the variations have a far higher frequency as well as for this reasons are less easy to conceptualize although uniformly infuriating.
Exceedingly often in these scenarios the rhythmic message, the pace of the most complicated musical plots is partially or completely lost, music is dull, hardly connecting and it seems that pointless, it does not make any widespread sense. An ingredient for austerity, the archetypal digital reverberation, in an expression…. In subordinate amounts, the impact beyond is intricate to conceptualize although jitter is still able to bring about anomalies reduction of the soundstage width as well as depth, requirement of spotlight, sometimes a veil on the melody. These outcomes are nevertheless far more intricate to trace back to jitter, as can be caused by many other variables, Dunn (2003: 20-22).
Quantization racket
Analog structures do not have discrete digital echelons in which the waves are encoded. As a result the inventive wave can be preserved to a precision restricted only by the intrinsic racket base as well as optimum signal echelon of the media as well as the replay apparatus. Thus the dynamic range of the structure. With digital structures, noise augmented due to quantization into discrete echelons is more audibly disturbing than the racket base in analog structures. This type of malfunction, sometimes known as granular or quantization distortion has been highlighted to as a blunder of some digital structures and footages. Knee & Hawksford (1995:3) sketched attention to the deficiencies in some early digital recordings where the digital release was said to be inferior to the analog version.
Surplus stipulations and energetic assortment
Certain differences do exist with regard to traits of analog as well as digital structures when elevated echelons waves are accessible, where there is the possibility that such waves could push the structures into overload. Elevated echelon waves, analog magnetic strips approaches saturation as well as elevated tempo response drops in proportion to diminutive frequency rejoinder. The audible impact of this can be reasonably unobjectionable (Elsea 1996:74). In comparison, some digital PCM recorders can illustrate non-benign traits in overload (Dunn 2003:65). The softness of analog ribbon clipping enhances a usable dynamic range that protracts beyond that of some PCM digital recorders.
Aliasing distortion
The mentioned disadvantages of digital audio systems have been the subject of discussion. With regard to aliasing distortion, Hawksford (1991:18) highlighted the advantages of digital converters which operate at higher than the Nyquist rate (i.e., over sampling converters). Using an over sampling design and a modulation scheme called sigma-delta modulation (SDM), analog anti-aliasing filters can effectively be replaced by a digital filter. This approach has several advantages. The digital filter can be made to have a near-ideal transfer function, with low in-band ripple, and no aging or thermal problems.
Frontage Panel 7 Band 12 Band
Two diverse front panels employed for the range do exist. Aesthetically they are rather analogous; nevertheless proximity inspection indicates that myriad disparities are present. The 7 BAND has all its elementary facades on the frontage, yet with little on the back. In contrast the features of a 12 BAND have been concentrated on the back hence creating more room for additional features on the front.Fundamentally the procedure is much uncomplicated. The wave stream darts from left towards right along the operation on the frontage plate. The rank of each operation is evidently visible by the corresponding controls and LEDs that also match the LEDS on the foot controller.
Audio compressor
The increase of the distinct phase virtual earth amp IC1 is resolute by the draw off-foundation f.e.t. resistance. The f.e.t V-1 attributes are linearised by resistors R1, R2 as well as R3. Control energy is sourced from the production wave by employing an accurate rectifier as well as max out detector, Ashihara, K. et al. (2005:50-54). Hit and crumble interludes are adjustable by resistors R4 as well as R5, in addition to the values indicated giving interlude constants of 1 and 517ms correspondingly. The compressor is enhanced to function as a conventional fixed-gain amplifier courtesy of the joint switch which runs a preset control power.
In the restrict mode, a 29dB amend in the contribution wave echelon produces a 9dB change in the output.
Figure 1. L. Mayes, Ipswich, Suffolk.
Linearization process explained
Hit interlude constant is measured by the product of R4 as well as the 4.7µF capacitor. When the input signal drops D1 develop into a turn around predisposed and the decay interlude constant id measured by R5. Mutually time interludes are an aspect of a compromise- the hit ought to be fast especially when the start of elevated amplitude waves are not to be congested until the increase is minimized and the oxidation should be fast enough to validate stumpy amplitude waves abruptly next to elevated amplitude ones that have to be given sufficient increase. Anomaly emerges after long interludes of quiet or low amplitude inputs- the subsequent elevated amplitude wave will achieve the full gain treatment and as such will initially overload the system and consequently malfunctions will be evident.
The reduction of R4 to zero could be the best remedial process that could be implemented, hence resulting in minimal attack interlude. The circuit is by no means hi-fi yet it is also significant for agc fastidiously in tape recording radio and wave proliferation where a wave’s large dynamic range require to be minimized, Ashihara, K. et al. (2005:50-54).
Introduce your vocal file to a selected track, and employ the strip silence operation in logic to preclude any unwanted hiss or noise. A stumpy Threshold configuration of about one percent might be required. In keeping the vocals resonating innate, turning up the hit and discharging the configurations is ideal. The track can be played through the make a note of any aberrant sounds or hiccups. This could be due to the audio being engraved when the waveform is not at a 0 passage. Whoosh right into the waveform and examine….if this persists, the audio should be stretched out the audio until the hiccup has vanished, Borwick, J. et al. (1994:481-88).
Currently Compression
Most lyrics will require a minimal compression, although it depends to the individual. Did the vocalist have a poor microphone system, implying that the loud bits are too loud while the soft bits are not that loud? As such, compressing the wave is much more needed. Nevertheless, if the dynamic range has to be maintained about the track, then additional of the little compression is imperative. Whatever to be added is depended on the Threshold as well as Ratio configurations. The Threshold establishes at what point (dB) the compressor will start wraps the wave as well as the Ratio by how much. Attempt a hit configuration of 3ms, and a discharge of 91ms. Fool around with diverse system forms accessible until you establish one that you are content with.
Hybrid systems
Whereas the term analog audio is most cases mean that the aural is illustrated using a uninterrupted interlude, unremitting amplitudes line of attack in both the media as well as the reproduction/recording structures, the terminology digital audio mean a distinct interlude, isolated amplitudes approaches, then methods of encoding audio fall somewhere flanking the two, e.g. unremitting epoch, disconnect dimensions as well as detached instances, uninterrupted levels, Borwick, J. et al. (1994:481-88). Whereas not as common as ‘untainted analog’ or pure digital methods, these scenario do happen in observation. Undeniably, all analog structures indicate separate trait at the minuscule extent, and thus asynchronously operated class-D amps even consciously integrate unremitting intermittent, separate amplitude blue prints. Unremitting amplitude, separate interlude configurations have also been employed in most early analog-to-digital adaptors, in the shape of model-and- clutch systems. The precinct is more so distorted by digital structures that statistically aim at analog like characteristics, and frequently making good use of stochastic dithering as well as noise seminal performance. Borwick, J. et al. (1994:481-88).
Whereas album records as well as common compact cassettes are analog media and employ quasi-linear mechanical encoding approaches regardless of the noticeable quantization or aliasing, there are analog non-linear structures that show effects analogous to those encountered on digital ones, this include aliasing and hard dynamic bases. Even though hybrid methods are normally more common in telecommunications structures than in consumer audio, their subsistence alone blurs the distinctive line that flanks various digital and analog structures, at least for what observe some of their supposed pros and cons, Ashihara, K. et al. (2005:50-54).
Discrete variation of Bass as well as Treble without Switches
The circuit to be illustrated is as the result of protracted evaluations of tone control systems of the unremitting compliant shapes, as well as present discrete control system of the unremittingly amendable style, and offers self-regulating control of low and treble reply by means of two potentiometers, minus the requirement for knobs to alter over from haul up to slash. Remarkable characteristics are wide-ranging hodgepodge of control accessible and the verity that a level retort is acquired equally with potentiometers at median configuration. Treble rejoinder arch’s are of a continuous profile, being distorted across the regularity federation when the control is operated, and there is pragmatically no propensity for the curvatures to roll out off towards the upper limit of the audio rang. The shape of the bass response curves, even though not invariable varies less than with most continuously adjustable systems, Ashihara, K. et al. (2005:50-54).
The practical earth conjecture
The presentation sketched above has been realized by the applicability of negative response system instead of the more usual passive form of structure that is desirable that the learner ought to be familiar with the pragmatic earth hypothesis as functional to response amps, before the functionality of the tone control circuit is considered in detail. Shared Treble Control System- In order to acquire treble pinch or engrave with single know control, the two systems just illustrated may be pooled by employing a nucleaus tapped potentiometer, this is accessible commercially in carbon track form at a price only slightly in excess of that for an ordinary potentiometer. The centre tap is earthed, on end of the element is integrated to the input workstation; the other end is connect to the valve anode as well as the slider is connect to the capacitor C, Sony Europe (2001:20-27).
Graphic Equalizer Switch & Sliders
When controlled, the explicit equalizer will be inserted into the signal chain. The Emerald LED exceeding the sliders illustrates its condition. This can be controlled by switching on and off by the use of the front panel switch as well as from the foot regulator. The graphic controller is the centre piece, jointly functionally as well as quite literally, of a conventional. Trace Elliot amps, so much so that the authentic name for every part of the range is derived from the number of frequency band within each one, i.e. 7 Band and 12 Band. Both forms are very powerful gadgets for monitoring the tone of your instrument. The blueprint of a Trace Elliot equalizer is very disparate from a specification echo support EQ, 15dB of cut boost is available for every band, although instead of being spread numerically over the tempo dimension they have been set at carefully selected tempos of fastidious application for low guitar, Ely, S. (1978:30-5).
As a result, there is more power in the subordinate concurrencies. Subsequent disparity is normally that every slider monitors a more expanse tempo dimension than normally found..
The blueprint disparities are in common with conventional Trace Elliot amps that are also part of the sound. With the monitors all set at 0dB, no alteration is made to the wave. Heartrending a slider up will progressively enhance, or boost’ the tempos centered on the tempo mentioned above the slider. Moving a slider beneath will enormously decrease or cut the tempos. Owing to flexibility and massive cut and boost available, it is significant to know how to get the best from the system, Stark, C. (1989:625).
References
Ashihara, K. et al. (2005). “Detection threshold for distortions due to jitter on digital audio”, Acoustical Science and Technology, Vol. 26 (2005) , No. 1 pp.50–54.
Blech, D. & Yang, M. (2004). “Perceptual Discrimination of Digital Coding Formats”, Audio Engineering Society Convention Paper 6086, 2004.
Borwick, J. et al. (1994). The Loudspeaker and Headphone Handbook, 2nd edition. Edited by John Borwick, with specialist contributors. Section 11.7 ‘Experimental Procedure’, by Floyd Toole, pages 481-488. Focal Press. ISBN 0 240 51371 1.
Croll, M. (1970). “Pulse Code Modulation for High Quality Sound Distribution: Quantizing Distortion at Very Low Signal Levels”, Research Department Report No. 1970/18, BBC.
Driscoll, R. (1980). Practical Hi-Fi Sound, ‘Analogue and digital’, pages 61–64); ‘The pick-up, arm and turntable’, pages 79–82). Hamlyn. ISBN 0 600 34627 7.
Dunn, J. (1998). “The benefits of 96 kHz sampling rate formats for those who cannot hear above 20 kHz”, Preprint 4734, presented at the 104th AES Convention, 1998.
Dunn, J. (2003). “Measurement Techniques for Digital Audio”, Audio Precision Application Note #5, Audio Precision, Inc. USA.
Elsea, P. (1996). “Analog Recording of Sound”. Electronic Music Studios at the University of California, Santa Cruz.
Ely, S. (1978). “Idle-channel noise in p.c.m. sound-signal systems”. BBC Research Department, Engineering Division. Pg 30-5
Greenfield, E. et al. (1986). The Penguin Guide to Compact Discs, Cassettes and LPs. Edited by Ivan March. Penguin Books, England.
Greenfield, E. et al. (1990). The Penguin Guide to Compact Discs. Edited by Ivan March. Preface, viii-ix. Penguin Books, England. ISBN 0 14 046887 0.
Hawksford, M. (1991). “Introduction to Digital Audio”, Images of Audio, Proceedings of the 10th International AES Conference, London, 1991.
Hawksford, M. (1995). “Bitstream versus PCM debate for high-density compact disc”, ARA/Meridian web page, 1995.
Hawksford, M. (2001). “SDM versus LPCM: The Debate Continues”, 110th AES Convention, paper 5397.
Hicks, C. (1995). “The Application of Dither and Noise-Shaping to Nyquist-Rate Digital Audio: an Introduction”, Communications and Signal Processing Group, Cambridge University Engineering Department, United Kingdom.
Jones, W. et al. (2003). “Testing Challenges in Personal Computer Audio Devices”. Paper presented at the 114th AES Convention. Audio Precision, Inc., USA.
Kaoru, A. & Shogo, K. (2001). “Detection threshold for tones above 22 kHz”, Audio Engineering Society Convention Paper 5401. Presented at the 110th Convention, 2001.
Knee, A. & Hawksford, M. (1995). “Evaluation of Digital Systems and Digital Recording Using Real Time Audio Data”. Paper for the 98th AES Convention, 1995, preprint 4003 (M-2).
Lesurf, J. “Analog or Digital?”, The Scots Guide to Electronics.
Libbey, T. “Digital versus analog: digital music on CD reigns as the industry standard”, Omni, 1995.
Lipshitz, S. “The Digital Challenge: A Report”, The BAS Speaker, 1984.
Lipshitz, S. (2005). “The Rise of Digital Audio: The Good, the Bad, and the Ugly”. Abstract of Heyser Memorial Lecture given by Prof. Stanley Lipshitz at the 118th AES Convention.
Liversidge, A. “Analog versus digital: has vinyl been wrongly dethroned by the music industry?”, Omni, 1995.
Manson, W. (1980). “Digital Sound: studio signal coding resolution for broadcasting”. BBC Research Department, Engineering Division.
Metzler, B. (2005). “The Audio Measurement Handbook”. Second edition for PDF. Audio Precision, USA.
Nishiguchi, T. et al. (2004). “Perceptual Discrimination between Musical Sounds with and without Very High Frequency Components”, NHK Laboratories Note No. 486, NHK (Japan Broadcasting Corporation).
Paul, J. “Last night a mix tape saved my life”, The Guardian, 2003.
Pozzoli, G. “DIGITabilis: crash course on digital audio interfaces. Part 1.4 – Enemy Interception. Effects of Jitter in Audio”, “TNT-Audio – online HiFi review”, 2005.
Pohlmann, K. (2005). Principles of Digital Audio 5th edn, McGraw-Hill Comp.
Rathmell, J. et al. (1997). “TDFD-based Measurement of Analog-to-Digital Converter Nonlinearity”, Journal of the Audio Engineering Society, Volume 45, Number 10, pp. 832–840; 1997.
Rumsey, F. & Watkinson, J. (1995). The Digital Interface Handbook, 2nd edition. Sections 2.5 and 6. Pages 37 and 154-160. Focal Press.
Sony Europe (2001). Digital Audio Technology 4th edn, edited by J. Maes & M. Vercammen. Focal Press. Pg:22-27
Stark, C. (1989). Encyclopædia Britannica, 15th edition, Volume 27, Macropaedia article ‘Sound’, section: ‘High-fidelity concepts and systems’, page 625.
Stuart, J. (n.d.). “Coding High Quality Digital Audio”. Meridian Audio Ltd, UK. Retrieved 9 March 2008. This article is substantially the same as Stuart’s 2004 JAES article “Coding for High-Resolution Audio Systems”, Journal of the Audio Engineering Society, Volume 52 Issue 3 pp. 117–144; 2004.
E-delivery of audio from the holdings of the Sound Archive will facilitate the New York Library in making an exceptional contribution to the latest training setting for NY higher education, and will offer pointers to the enormous unexploited resources that lie beyond the very minute but convincing collection of tapes to be integrated in this project, hence contributing to the New York Library vision of assisting individuals in advancing knowledge with a view of enriching lives.
Project objective
The ASRP will offer unlimited access, at the level of provision, to the program contents for the higher education industries in the United States. Access will as well be offered to American library facilities. Free access will in addition be provided over the e-site to a percentage of the content theme based on eligibility status. The ASRP will offer 10 Content Products representing an expansive variety of Sound Record holdings.
Project scope
Included in the scope
Comment
Archival contents of agreed material resources
e-copies for streaming and access to agreed material resources
Extranet access to the agree material results to the higher education community
Resource identification and procedural services
Excluded from the scope
Website model
The model will be developed at project position. The ASRP will, however, update the strategies.
Web-hosting and user interface
This will be determined at program position, with contribution from the program.
Project deliverable
Project phase
Deliverable
comment
Expected date
Project Brief
Project start
Project documentation
02.10.2012
Project start
Business case
Revise regularly
05.10.2012
Project start + Digital production phase
Project
The project will be revised when suppliers are selected
12.10.2012
Project start
Risk log
7-day updates
Project start
Quality plan
To be revised during negotiation with the suppliers
Updated regularly
Project start
Dissemination plan
For both intrinsic and extrinsic purposes
15.10.2012
Project closure
Project assessment report
To Joint Information system Committee and the NY Library
Handling policies for non-sound aspects in transmission-streams.
This ASRP’s packages will have a functional influence on the following sections:
NY library section
Impact
The NYL audio record’s technical service
On the administration of digitization agreements
The audio record
The program will update organizational strategy creation for maintenance and access.
EIS
Digital Content Administration Model (DCAM) program. The project will update DCAM as to the specifications for digital sound storage.
EIS
Digitization approach
Scholarship & collection
The creation of new electronic specifications for sound-based data and the encoding of sound contents.
Dependencies
Dependencies
Remarks
NY library project team
The final web-hosting feedback for this program and in the future the sustainability of a NY library system.
The Joint Information System Committee’s user specification register
The processes are planned to be finished by 20 February 2013.
The NY library’s system to deal with user specifications.
It ought to be recognized that the NY library is still shaping its thinking concerning how far to pursue the idea of offering informative package, in support of a specific digital product. Generally the library is in the idea of provision of timely digital content.
The incorporation of digital system in the NY library.
This will influence long-lived sustainability.
NY library website system
Cross-functionality concerns (NY library)
Project approach
Substantial study has been carried out with a view of proposing solution to the JISC for selection and for the purpose of developing a proposal for financing from the JISC and for the content collection procedure.
Content has been approved by a team of professionals and the Verification Template approved by the SRO. The final selection of content (i.e. ten Content Products – 5 from 15 by NY Library and five from the remaining ten by the higher education community) will be submitted to the Program Panel for approval.
A sample will allow response on the provision of recorded content over the e-site and will result in the establishment of user interfaces. The trial product is to be financed through The JISC under a separate allotment of financing and will be administered at Digitization Program position with inputs from the Archival Audio Recordings Project (Public Procurement Directorate, 2008).
During the Program Initiation phase the Program Team will be requested to approve the Program Document and the Program Initiation Brief. The two documents to be evaluated by CPO.
The sample digitization assignment will be conducted by the program group and will appear as 600 second prototypes across all the content products. Such packages will be examined through a Beta interface framework which will be based on usability tests. This procedure is to be recorded in the e-site register.
The verification of the e-site is subject to acceptance by the board. Suggestions for validation will be forwarded to the Program Team for approval.
During the Audio Processing phase, the Program Team will be requested to approve an Audio Template. This matrix will be utilized in communicating the development of content to the Program Team. It ought to be approved at the closure of the program by the Program Manager and will indicate that all agreed materials have been digitized.
The e-site register ought to as well be verified at the end of this phase demonstrating that all e-site deliverables are integrated based on the strategy outlined in the Website Strategy Brief.
The program’s results will be communicated to the Sponsor as per the original dissemination plan. The final brief will be determined by the NY Library and the Sponsor once the suppliers have been selected and a final implementation timetable written (Islam & Bhuiyan, 2011). Program assessment will be carried out by a self-governing evaluator. The report to be finalized one month prior to end of the program to enable remedial actions if necessary.
Fit with Organizational Strategic Plan
The New York Library was approached by the JISC to tender for financing for digitization programs as part of the Joint Information System Committee availability and archiving strategic plan. This chance emerged from an intrinsic JISC financing avenue known as the Comprehensive Spending Review (CSR).
This program is one of five programs being financed via the CSR Digitization program. The project as well involves the NY Library’s 19th century magazine program. Joint Information System Committee’s choice of the Audio Archiving proposal indicates the increasing attention to digital content from the United States educational community.
This solution was adopted by the NY Library as an excellent fit with its expansion plan for the electronic archiving of its library materials.
Anticipated benefits
The ASR program will make an important contribution to the adoption of the New York Library’s goal of increasing access to its materials. It In addition reacts to the demands from the U.S. higher education sector, articulated via the Joint Information System Committee, for the access of the digital content with a view of supporting schooling. The project will attain this by allowing:
Unlimited access to an assortment of audio tapes and related pictures characterized by 10 content products derived from the selections of the NY Library Audio Records, comprising verbal history consultations, different genres of literature, radio adverts and music. The collections will be authorized for learning purposes and delivered to educational consumers as primary study material.
Unlimited access to an extensive variety of exceptional training and/or learning resources previously handy only in UK-based classrooms.
The capability of integrating corresponding and explicit academic data stored in the form of electronic records along with the collection of reference materials usually accessible for research, training and e-learning.
In the future the lessons learned from this program will facilitate the Audio Record to enhance availability of its contents and to deliver on major organization objectives.
Cost estimates/cost benefit analysis
Action
Hrs
Days (8 hour/day)
Costs ($1500/day)
Remark (s)
Development
global
40
5
7500
Template & parts
600
75
112500
Customer
-200
-25
-37500
Subtotal/day
440
55
82500
HMTL incorporation (+15%)
66
8.25
12375
Multi-language ability (+15%)
66
8.25
12375
Subtotal
572
71.5
107250
OS service
48
6
9000
migration instruments
48
6
9000
Subtotal
668
83.5
125250
prototype testing/QA
100.2
21.53
18788
Clarification during establishment
57.85
7.45
12300
Documentation (+5%)
33
4
6325
Total development
859.05
116.48
162663
Testing
Basic test (user)
Functional test
80
9
14780
Debugging (=15%*development)
128.86
17.47
24399.45
Total testing
208.8575
26.472
39179.45
Deployment
8.45
2
1600
system requirement
8.45
2
1600
system tool
8.45
2
1600
testing environment
8.45
2
1600
processing environment
8.45
2
1600
website server/installation
8.45
2
1600
user definition
8.45
2
1600
Total Deployment
59.15
14
11200
Summary
Subtotal (development/test/deployment)
1127.058
156.952
213042.45
Program administration (+20%)
225.4115
31.3904
42608.49
Total
1352.47
188.34
255650.94
Tax (+15%)
270.49
37.67
51130.19
Total
1622.96
226.01
255650.94
Special fund sources
The Joint Information System Committee has given simply over a million pounds to The NY Library for the ASR project and budget allocations have been approved. Any additional item requires authorization by the project committee that this addition is important.
NY Library responsibility systems to be utilized for intrinsic monetary controls. Any deliverable agreed between the Joint Information System Committee and the Library is to be delivered as per the timescale.
Six monthly financial reports to be submitted to the Joint Information System Committee indicating expenditure alongside estimated budget.
All program group economic spending to be documented by the program assistant staff and checked by the program group head and program manager. Program manager to offer an exemption report to the program committee in case of an estimate of expected budget exemptions.
Risks
The preservation of the risk record is the role of the program leader and its analysis that of the program committee. The rise of risks is to be carried out in consultation between the program committee and the program leader. Input from the program group will be dealt with firstly via the revising of the Issue Record.
Any fresh risk or emerging concerns will then be integrated into the Risk Record for analysis by the program board and the project committee. As identified risks my turn out to be issues, the issue record, in turn, will be revised with a view of reflecting these risks (Islam & Bhuiyan, 2011).
The major risks comprise erroneous cost approximations at the proposal level, ambiguity at program initiation period about website-hosting options and expected costs, judicious approval of content collection, and the challenges and cost of IP authorization. The administration of such uncertainties is supported by an inclusive risk record, occasionally handled by exemption to the program committee for corrective measure where required (“Project Origination”, 2001).
Summary of program schedule
The program begins with the program manager on the ground at the beginning of October 2012, and will continue for the next 2 years until the end of October 2014, at which point the electronic-based services will be working with all ten audio products available.
Constraints and assumptions
Constraint 1
The project budgetary allocation is approved by the NY library and the Joint Information System Committee and is not likely to be adjusted
Constraint 2
The project completion date is 30 November 2012
Constraint 3
Access is mainly to higher education community
Constraint 4
Joint Information System Committee usability testing will prevent the establishment of the user platform
Constraint 5
NY library accessibility policies will be followed as required by the JISC
Constraint 6
The acquisition of digitization services is subject to U.S. practices enacted in U.S. regulation and procurement will be subject to a Negotiated Process (“Project Initiation”, 2001).
Assumption 1
That a right is attainable for all chosen content products.
Assumption 2
Access to any library sound content will be available in electronic form on NY Library facilities provided this is program budget-based. If it is not then this requirement will be beyond the scope of this program.
Assumption 3
All suppliers can account for approximately 85% of the work in 2 years.
Assumption 4
Any production work will be carried out by the program group and will only make minimum challenges to the technical team.
Organizational structure
Job title
Role
Program Board
Approval of the program deliverable
Top Responsible Officer
General accountability for the effective completion of the program
Program Director
Daily program administration. Reporting to the program board and the Joint Information System Committee
Program team Manager
Daily administration of the program team comprising quality assurance and negotiation concerning IP issues. Reporting to the program director
Program Team
Processing and advancement of chosen sound package and integration of chosen information aspects.
JISC Person
The project leader representing Joint Information System Committee on the program panel
Higher Education Representative
Offers HE user contribution, participating in program board activities as necessary
NY Library Digitization Program Manager
Offers leadership on NY library digitization program concerns and general administration support to the program.
Corporate Program Officer
Offers consultation comprising NY library program administration techniques.
Procurement Department
Accountable for the implementation of U.S. procurement practices (Public Procurement Directorate, 2008).
Website Service Provision Department
Accountable for the provision of the consumer platform to package. Updates the website approach for the program.
Communication plan
Stakeholder analysis
Stakeholder
Interest
Communication need
NY library management
All New York library schedule of work
Progress report.
JISC
Program completion
Weekly consultation with the program director and project management report.
Director of scholarships and collection
Program sponsor
Progress report and program communication.
Head of New York collection
Program completion
Program board representative. Needs Progress report and program management report.
NY Library digitization project manager
Project completion
Program board representative. Needs Progress report and program management report.
Communication strategy
Negotiations based on timetabling of project activities with the suppliers will automatically update the extrinsic communication of information concerning major deliverables and extrinsic communication plans will be agreed once the suppliers have been selected. The completion date for such selection is 18/10/2012.
It is proposed that November would be an excellent period for since there are frequently breaks in press releases worthy legends at such period (CHUA, 2001).
The program committee comprises all NY Library units with stakes in the program and the Joint Information System Committee via their member and the higher education representative.
Regular bulletins will as well guarantee that development is communicated more broadly in the NY Library and to the Joint Information System Committee.
A printed timetable of actions and events will be developed alongside the Joint Information System Committee Dissemination Strategy (Islam & Bhuiyan, 2011).
Project quality administration strategy
Quality will be determined against NY Library audio record specifications for digital storage and as well against all other specifications agreed with the suppliers.
Data will be certified by sampling for any error and lacking explanations alongside the NY Library Plan.
Agreement on any standard will update the discussion with the suppliers and this will lead to a cycle of benchmarking process alongside which quality will be determined via sampling by the program group.
Quality concerns are to be integrated in the program team’s standard report to the program committee.
Completion of quality assurance schedules to be disseminated to the program committee via the digitization template (Islam & Bhuiyan, 2011).
Acceptance criteria
The functional surrender of the website and hosting criteria will be submitted to the New York Library thirty six months after the program closure. In the provisional functional administration will be carried out by an independent supplier based on an administration SLA.
Deliverable
Acceptance criteria
Remark
Program documentation
Consistency for purpose
Documentation managed by JISC
10 packages
Tested alongside quality assurance specifications captured in digitization template
Over 12 thousand access and printable documents
Tested alongside quality assurance specifications captured in digitization template
The preservation and provision alternative
Tested at program stage alongside U.S. specifications
Sustainability plan
Any documentary copy will be treated as a component of audio record holdings before final integration into DCAM.
All available documents will be attached to the agreed web-hosting platform.
Function and preservation will be the role of the NY facility for thirty six months from November 2014. An Entire Service Administration Agreement will be integrated with a view of ensuring this.
Fresh Service Level Agreements will then be integrated and form the foundation of the agreement for continuing website service for this program. This will be preserved for the thirty six months after project closure.
A sustainability strategy will be developed for the integration of the contents into the NY Facility DCAM by 2013.
References
CHUA, P. (2001). The initiation, organization and logistics of part-time final-year projects. International Journal of Engineering Education, 17(3), 248-254.
Islam, S. & Bhuiyan, U. (2011). The association between project success and project initiation phase: A study on some selected projects in Bangladesh. European Journal of Business and Management, 3(12), 60-68.
Public Procurement Directorate. (2008). Public procurement best practice guide (no. 1.1). Canberra, Australia: Author.
Project Initiation. (2001). In D. Chan (Ed.) NYS Project Management Guidebook (pp. 51-126). New York: Information Technology Press.
Project Origination. (2001). In D. Chan (Ed.) NYS Project Management Guidebook (pp. 21-45). New York: Information Technology Press.
An examination on the behavior of shrimps and crabs when exposed to sound within an enclosed environment.
Problem Statement: Do sound waves, when consistently projected towards invertebrates within an enclosed environment, result in their avoidance of that particular area?
Relevance of your testable question
As humanity continues to expand its activity to encompass more and more areas around the world the result has been a subsequent expansion of activity into previously secluded habitats of various marine organisms. With this expansion has of course come the daily sounds associated with human activity such as the sounds of ship engines, the noise heard during the construction of an offshore oil platform or even the regular sound created by people within a general area. The inherent problem with this sound is that it is a form of stressor, namely, an outside factor that affects the ability of marine organisms to continue with their normal activities due to the effect of this outside form of interference.
The inherent problem such a concept is that in most cases where a stressor is consistently applied to a particular environment the result usually results in marine organisms, especially invertebrates, leaving the immediate area. In the case of this particular study what will be examined is if sound can act as an effective stressor for invertebrates to actually cause them to avoid a particular area. It is believed that the results of this study will show that the sounds created by normal human activity within a given area cause marine organisms to avoid it thus precipitating a slow deterioration of the environmental balance within a given area.
Literature review
Marine invertebrates do not actually “hear” in the conventional sense, they lack the external sensory organs commonly associated with hearing to actually hear in the same way that various other species do. Various studies do show that they do respond to sound, as such this indicates that there must be some form of “hearing” taking place that causes them to perceive the sound itself. An examination of the interior workings of marine invertebrates shows that they possess organs known as chordotonal organs which act as a form of internal “mechanoreceptor”.
What this means is that these particular organs sense cues normally associated with sound such as vibration since these organs are also utilized as a means of detecting various forms of pressure, movement and tension. The perceived reaction of invertebrates to sound is the result of them perceiving the sound as an external reaction caused by either a predator, prey or natural occurrence within the environment. This triggers a particular response to either hide or investigate the source of the sound. It must be noted that in various cases it has been shown that the higher the decibel level of the noise produced the greater the tendency for invertebrates to avoid the sound due to an apparent association with either a large predator or a sudden natural calamity.
Experiments conducted by Boehlert and Gill investigating the effect of sounds on marine organisms within a given area reveal that on average marine organisms that react to large noises through avoidance tend to avoid a certain area depending on the given noise intensity threshold produced (Boehlert Gill, 2010). Similarly, Gero Vella in his experiments involving the effect offshore wind farms have on local marine environments show that certain decibel levels do indeed affect marine organisms, especially invertebrates, wherein decibel levels over 180db or more within a given area are sufficient to scare away certain types of marine organisms within a small enclosed area (Vella, N.I.).
Based on this it can be seen that depending on the level of sound produced marine organisms can and will avoid a certain area due to their instinctual “flight” mechanism (referring to the fight or flight mechanism) associated with large vibrations created by distinct sounds being associated with a large predator or natural event. It must be noted that while such experiments show that marine invertebrates do avoid sound none of them mention what happens when the sound is removed and reintroduced at a given rate.
Experimental design
Step in the Experimental Procedure
Based on the lack of data regarding what might occur when sound is introduced and then taken away at a given rate what this experiment will seek to accomplish is to examine the behavioral reactions of various invertebrate species when sound is introduced into an enclosed environment at different intervals.
Step 1
Various live invertebrate species need to be obtained to accomplish the results of the experiment. Local fish markets within the general area have live specimens of crabs and certain types of shrimp available that can be utilized for this experiment. In order to better recreate their natural environment seawater will need to be procured which is also available at the fish market since this is what is used to keep the specimens alive during transport.
Step 2
A large pail roughly 3 feet tall and with a diameter of 15 inches will be used as the recreation for the natural environment. To ensure that the only external factor in this experiment will be sound waves, the exterior of the pail will be wrapped in black felt paper and the top covered with a lid in order to ensure that the only influencing factor in this experiment will be sound.
Step 3
The seawater along with the live shrimp and crabs will be placed into the pail and given at least 2 hours recuperation time to ensure that they will be ready enough to endure the experiment.
Step 4
Two large speakers will be procured and used to simulate the effect of sound in their natural environment. The speakers will be placed in alternating areas throughout the outside of the pail in order to simulate the effects of sound coming from different directions in the natural habitat of the shrimps and crabs.
Step 5
The type of music for this particular experiment to simulate the sounds of natural human activity within a given area will be Rammstein by the band “Rammstein” as shown in the following YouTube.
The reason why this particular song was chosen was due to the level of acoustic vibrations caused by the speakers when it was tested early on.
Step 6
After the Speakers have been placed and the experimental subjects have been given enough rest the experiment will begin. The speakers will be played at a volume of 180 decibels for approximately 5 minutes in one location then turned off and transferred for another location and played again for another 5 minutes. During the end of each alteration, the lid of the experimental environment shall be lifted to observe the behavior of the crabs and shrimps inside to see whether the music has any discernable effect on their behavior. This process will continue until 4 different directions have been achieved and their results recorded.
The reason why this particular experimental design was chosen above all others was because it was the only method of feasibly recreating the needed environmental conditions needed to effectively examine the behavior of invertebrates when sound is introduced into their natural habitat. Attempting to do so without the aid of an enclosed and controllable environment could fail the experiment due to numerous outside various possible contaminating the results.
Measuring the Reaction of the Invertebrates
In this particular experiment is assumed that the behavioral response of the various organisms utilized in this experiment will take the form of movement. With one of the dependent variables being the consistently in sound levels (160 DBS), it is assumed that by exposing the subjects to the same sound yet at varying directions the result will be that the subjects will either move toward or avoid the sound by moving away. In order to prove that the subjects moved a ruler will be placed at the bottom of the container to measure the degree by which the subjects actually moved. The experimental procedure measurement will proceed by first taking note of the current location of the subjects before the music is applied then taking note of their location after it is applied. The resulting movement either away or toward the music will serve as the experimental results.
Tools utilized in the Experiment
The tools utilized in this particular experiment are nothing more than 2 common rulers. One to measure the rate of movement from North to south while the other measures movement from East to West. The degree of movement will be measured utilizing inches since it was decided that centimeters would be far too troublesome to determine. Other units of measure were not needed since throughout the experimental procedure all factors were controlled such as the level of sound and the amount of outside environmental factors.
3. The controlled variables in this particular experiment consisted of the environment that the experimental subjects were in, the temperature of the water which remained at room temperature and the level of sound coming from the speakers which remained at a constant 160 DBS throughout the experiment. The dependent variable in this particular study consisted of the distance traveled by the experimental subjects in each direction as the sound was applied. While the independent variable itself is the different rates of distance traveled by the experimental subjects since each responded in a slightly different way (as indicated by distance traveled) in some instances throughout the data gathering step of the experiment.
It must be noted that while this experiment could utilize a method of experimentation wherein different sound levels can be used the fact remains that based on the data of Vella sounds lower than 180 dB probably could not sufficiently produce enough of a reaction in invertebrate species to cause them to produce any behavioral changes. It is due to this that 160 DBS was the chosen sound level for the dependent variable since by put the speakers close enough to the pail this should cause enough transference from sounds waves to a vibrational reaction in the water to elicit a reaction from the experimental subjects.
4. In order to reduce threats to the internal validity of the study, certain precautions will be undertaken to ensure that the only means of interaction between the test subjects and the outside world will just be the sounds from the speakers. First off light as a method of eliciting a reaction will be blocked off completely from the pail using black felt paper. Secondly, a glass lid will be placed over the pail with black felt paper covering the other side to simulate total darkness. It is in this particular environment where nearly all external stimuli are blocked off that the experiment will proceed with the introduction of the external factor of sound. It is expected that utilizing this particular method all external threats to internal validity will be prevented thus ensuring the integrity of the experiment.
Hypothesis
It is the hypothesis of this experiment that by introducing sound waves at different intervals from various directions that the experimental subjects of this experiment will move away from the sound. This particular hypothesis is based on the data Boehlert and Gill investigated regarding the effect of sounds on marine organisms wherein the introduction of sound within a given area can cause enough of a vibrational reaction to actually cause various marine animals to move away due to the “flight” instinct regarding the connection between large vibrations and danger.
Process of Data Collection
Experiment Photos
(Seen here are the live crabs without the shrimp and seawater placed in)
During the experiment, it was discovered that the bucket utilized was too small for the crabs or shrimp to effectively move around in due to the lack of space. An alternative container had to be utilized that had enough space for the various experimental subjects to move around in. A small used red container was found and covered with a plastic bag since the felt paper bought turned out to be unnecessary for both the bucket and the container.
(New Experimental Container)(Decibel Level Guide for Experiment)
The process of data collection for the experiment was relatively straightforward. On each corner of the box, music was played at approximately 160 decibels as close to the box as possible. Before the start of each experiment, a stick was used to move the experimental subjects close enough to the side of the box where the music was to be played and then the experiment was covered. The music was played for 5 minutes on each side and the distance traveled by the experimental subjects from the side of the box where the music was played was measured by a ruler placed inside the box.
Methods of Data Collection
The quantitative data were collected utilizing a ruler that placed in the container beforehand to measure the degree of distance traveled between the side where the music was playing and the rate in which the experimental subjects traveled away from it. No complicated methods of measurement were required since this was a relatively straightforward process examining the “flight” mechanism of invertebrates when exposed to relatively high sounds resulting in vibrations in the water.
Results of Experiment
Decibel Level
Crabs (Inches Moved)
Shrimp (Inches Moved)
160db (Left Side)
4 Inches to the right
2 Inches to the right
160db (Right Side)
2 inches to the left
2 inches to the left
160db (Top)
3 inches down
1 inch down
160db (Bottom)
2 inches up
1 inch up
Experimental Subject Movement Chart
Experiment Results
The result of the experiment shows that utilizing 160 decibels of sound over a prolonged period placed directly against a container does indeed cause enough of a vibrational reaction in the water over a prolonged period to elicit a reaction from the experimental subjects. The results of the experiment clearly show that over 5 minutes the shrimp and crabs moved away from the jarring sound which indicates that invertebrate organisms do move away from areas where there is a prolonged and persistent level of a particular high decibel level.
While this experiment did not intend to measure the level of sensitivity of the organisms the results do show that due to the distance traveled crabs are more sensitive to shrimp when it comes to a reaction to sound. This may be due to the more complex sensory organs in crabs as compared to shrimps and as such indicates that more complex organisms will have a higher tendency to move away from areas of high decibel levels as compared to other creatures.
Conclusion
The results of the experiment show that by introducing a high level of constant sound in a particular direction the resulting behavioral response from invertebrate organisms is to move away from the loud noise. This is shown by the varying directions shown in the experimental results where it can be seen that a certain degree of movement away from the offending sound can be noticed. This result supports the hypothesis based on the data provided by Boehlert and Gill that sound waves have a distinct effect on marine organisms, particularly invertebrates, where the characteristic vibrations produced are attributed to danger in the surrounding environment. The resulting movement away from the sound as seen in the experiment is thus attributed to the “flight” mechanism in marine invertebrates.
The experimental design was a key factor in proving this hypothesis since through the utilization of loudspeakers as a method of replicating the sounds of persistent human activity it can be seen that areas, where there are large concentrations of structures being built such as offshore oil platforms or areas, traversed regularly by ships causes a sufficient level of high decibel sound that this in effect causes invertebrate creatures to leave the area causing a subsequent change in the local ecosystem. In order to verify the results of this experiment all a person would need to do is obtain the same experimental subjects and subject them to the same decibel level utilized in this experiment over the same prolonged period to replicate the results. It is expected that through a replication of the features of the experiment wherein the same result is produced this would prove the validity of the results of the experiment and further cement the findings of this study.
Reference List
Boehlert, G, & Gill, A. (2010). The environmental and ecological effects of ocean renewable energy. Oceanography, 23(2), 68-79.
Vella, G. (N.I.). The environmental impacts of offshore wind generation. Center for Marine and Coastal Studies, 126 – 129.
