how acoustic engineers improve the sound quality of an auditorium?
362934
They control the Behavior of Sound Waves(Reflection, Refraction, and Diffraction)\n\nreflection of sound waves in auditoriums and concert halls do not always lead to displeasing results, especially if the reflections are designed right. Smooth walls have a tendency to direct sound waves in a specific direction. Subsequently the use of smooth walls in an auditorium will cause spectators to receive a large amount of sound from one location along the wall; there would be only one possible path by which sound waves could travel from the speakers to the listener. The auditorium would not seem to be as lively and full of sound. Rough walls tend to diffuse sound, reflecting it in a variety of directions. This allows a spectator to perceive sounds from every part of the room, making it seem lively and full. For this reason, auditorium and concert hall designers prefer construction materials which are rough rather than smooth.
534881
microphone and speakers
636030
Could be better speakers if that were the case. Otherwise the IBM had a better sound card. Your sound card and your speakers will both affect how good your speakers sound although if you want louder, go with bigger speakers.\n\nHere are some good ones to choose from...
509166
The answer is probably a yes, but the exact situation will change the answer.\n\nIf you are talking about background noise between the voice phrases you want to hear (Called the Noise Floor), this can be fixed fairly easily if the noise floor isnt too high. What you would do is use a device called an Expander/Gate. A Gate basically mutes (True muting or reducing volume by -60db or more) the audio when the volume goes below a certain level (Basically when you're not talking). If the noise isnt too loud, it will usually be masked by the person talking.\n\nAn Expander is like a more subtle version of a gate. Rather than killing the audio completely, it increases the difference of volume (expands) between loud and soft parts of the audio.\n\nMost video recorders reduce the dynamic range of audio using a proccess called compression. This has the effect of making soft parts of the audio intelligable, while making the loud parts not be over bearing. The problem however, is that this proccess increases the noise floor. I would hazard to guess that this is a major part of the problem you are experiencing if the audio you are using is running through the video recorder.\n\nIf this is the case, a gate will have a hard time functioning because the difference between the loud and soft parts of the audio will be hard for the device to detect (as the difference has been reduced). To compensate, stick an expander before the gate. This will add some dynamic range back to the audio, making it easier for a gate to work. This entirely depends on how much the dynamic range has been compressed... the more it has, the less effective these devices will work.\n\nThe operation of expanders/gates can be tricky. Threshold, attack, release, sidechain/key inputs, and ratio control operation are not immediatly obvious to someone who hasnt used an audio device with these functions on it. However, there happens to be tons of info on the net about these devices.\n\nRegarding using equalizers... they are mostly useful for background hums rather than broadband noise. If the noise sounds like noise (Like a "shhhhh", people talking in the background, wind, etc), EQs will not be very effective. If the sound is more like a buzz or humm or whine (Like an electric motor), then EQ will do very well. Look for a "parateric" EQ when doing this. There are tons of software that has this. Parametric EQs will allow you to not only center on the exact frequency, but the bandwith of the EQ can be set to be very narrow. The more narrow the bandwidth is, the less you effect your original sound. Also, you can kill pretty much all frequencies below 60-80hz (depending on the voice) without have a noticeable impact on the the voice. This will reduce some of the low end rumble in the audio.\n\nIf neither of these options work to satisfaction, one last option is a sampling noise reduction system. Not many programs have this... a free one would be GoldWave (Not very high quality, but still works in moderation). What it does is it takes a sample of the noise floor, notes the frequency response of the noise (called a stamp or print), and then reduces these frequencies from the audio. Used to the extreme, it will produce VERY weird artifacts... used well, this kind of noise reduction is currently the most effective in the world. If you can find software that does this dynamically (doesnt take just one "stamp"), the results can seem like magic.\n\nAll of these options can be attained through hardware, stand alone hardware, or via plugins.\n\nHope that helps.
403045
Have you thought of a ribbon mic?  The problem with harps is that their sound develops a couple of feet away from the soundboard, so its difficult to keep out bleeding while still getting optimal sound.  Good luck.
716327
Goel has given you the correct answer.\n\nIf there is no damping of sound waves, sound once produced will last for ever. \n\nThe intensity of a sound wave having a frequency of 100 cycles per second, decreases by a factor of 1.015.\n\nThe intensity of a sound wave having frequency of 20, 000 cycles per second, decreases by a factor of 10^274.\n\nUltra sonic vibrations are damped so rapidly that their transmission over a distance of more than several hundred meters is completely impractical.\n\nWhen a source of sound in an auditorium is cut off the sound energy in the room does not vanish immediately. It takes certain time. This time depends upon the absorption of sound energy by the materials present in the room.\n\nThe persistence of sound in a room even after the source has been cut off is called reverberation.\n\nThe reverberation time is the time for the original intensity to fall to the audible limit.\n\nIf the reverberation time is below 0.5 second the speech is not heard in a room.\nIf the reverberation time is between 0.5 second and 3 second the speech is heard clearly.  \n \nIf the reverberation time is more than 3 second and if the time interval between consecutive reflections is less than 1/15 of a second, the sound is unclear.\n\nIf the time interval between consecutive reflections is more than 1/15 of a second, clear sound (echo) is heard.
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depends on what you want to do with the sound, but generally they use geometry to create sound friendly environments. (feedback reduction,  good resonance reverb, noise reduction, and others)
26520
it helps you to get rid of the noise .
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Acoustical engineering is the branch of engineering dealing with sound and vibration. It is closely related to acoustics, the science of sound and vibration. Acoustical engineers are typically concerned with:\n\nhow to reduce unwanted sounds \nhow to make useful sounds \nusing sound as an indication of some other physical property \nThe art of reducing unwanted sounds is called noise control. Noise control engineers work with engineers in most industries to ensure that their products and processes are quiet.\n\nThe art of producing useful sounds includes the use of ultrasound for medical diagnosis, sonar and of course sound reproduction.\n\nA separate and related discipline, audio engineering, is the art of recording and reproducing speech and music for human consumption.\n\nRetrieved from "http://en.wikipedia.org/wiki/Acoustical_engineering"
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Audio engineering is a part of audio science dealing with the recording and reproduction of sound through mechanical and electronic means. The field of audio engineering draws on many disciplines, including electrical engineering, acoustics, psychoacoustics, and music. Unlike acoustical engineering, audio engineering generally does not deal with noise control or acoustical design. Much of audio engineering is also used in broadcast engineering.
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In the studio, everyone is a good singer.\n\nWith enough takes and processing, a skilled engineer can make anybody sound great.  This is often necessary when actors take up music.
1005476
Try going to Control Panel and adjust the sound settings
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>Acoustics is the study of sound: the production, detection, and propagation of sound waves, and the absorption and reflection of sound. It includes the study of how electrical signals are converted into mechanical signals, as in loudspeakers, and the converse, as in microphones; also, how sound is produced in musical instruments, and perceived by audiences in concert halls; the protection of workers from damaging levels of sound; and the use of detection techniques, such as sonar and ultrasonic scanning.
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As I understand it, it is sound that is broadcast over a PA system (Like in an office setting), that actually neutralizes the noise that is ocurring in said office environment. It doesn't drown it out, in fact one doesn't even notice it. It is projected at a frequency that obliterates the frequency of the noise already ocurring in the room. Its supposed to help in the work environment.
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All of these people worked on models to aid in acoustical design of ampitheatres (or any space where clearly hearing sound is important).  The three equations that you listed all take the same general form, and the latter two are subsequent improvements in Sabine's original work.\n\n"W.C. Sabine carried out a considerable amount of research in this area and arrived at an empirical relationship between the volume of an auditorium, the amount of absorptive material within it and a quantity which he called the Reverberation Time (RT). \n\nAs defined by Sabine, the RT is the time taken for a continuous sound within a room to decay by 60 dB after being abruptly switched off and is given by:\n\nRT = (0.161V) / A\n\nwhere V is the volume of the enclosure (m³) and A is the total absorption within the enclosure (sabine)."\n\n\n"Several different approaches have been used to derive equations which give values of reverberation time in better agreement with measured results from less reverberant rooms. One of these, the Norris-Eyring formula, assumes an intermittent decay with the arrival of fewer and fewer reflections. This gives the following formula:"\n\nRT = (0.161V) / (-S ln(1-a))\n\n"where S is the total surface area (m²) and a becomes the average absorption coefficient."\n\n"When the materials of a room have a wide variety of absorption coefficients, the best predictions are obtained by the Millington-Sette equation. This is simply a matter of substituting an effective absorption coefficient ae = -ln(1-ai) into Sabine's equation to give:"\n\nRT = (0.161V) / (sum(-Si ln (1-Ai))\n\nwhere Si is the surface area of the ith material, and Ai is its actual absorption coefficient.
201780
Unless the professor has special equipment offered then you probably aren't going to get very good sound quality. However, if you sit in the right place you should be able to get a legible quality through the idea you suggested. Just move your seats around some until you find a good spot. Also, make sure you ask your professor about recording their class, some of them could have very helpful things that they themselves use.
511133
Audio engineer...his company.installs multi-million dollar sound systems into theatres, complexes, arenas, etc. Lots of travelling to all countries..It's fun!