Audio video conferencing is a shorthand way of describing a virtual conference where no one has to leave her office to meet up with the rest of the group. A virtual conference may use telephones, televisions, computers, conferencing software, collaborative software, file sharing, headsets, the Internet, or any combination of these equipments and tools. Slide presentation, streaming audio, and document sharing are some of the extras you can incorporate into this conferencing method. The ultimate goal is to make the meeting as realistic as possible, where everyone can see each other, hear each other, view and modify projects, and so on – all at a cost that eventually pays for itself in travel savings.
The audio video industry has yet to work out all of the kinks. As you can imagine, there can sometimes be technical difficulties in which lines can get torn down in a storm, computers and other equipment can malfunction, and so on. Furthermore, audio video conferencing doesn’t work very well for complicated communication techniques, such as those used in closing a sales deal. Audio video conferencing lends a rather impersonal air to meetings, and socializing and bonding don’t take place as they would in the real world.
You can set up for audio video conferencing in a number of different ways. The older method, still great for a wide variety of teleconferencing purposes, is to have one group in one room, another group in another room (perhaps in another city or country), and have them talk to each other. Each of the two groups utilizes a television, a video camera mounted on the television, a speakerphone, and perhaps some file sharing software for collaboration on computer projects. The newer style is online audio video conferencing, wherein each participant has her very own station in her very own office. This utilizes a computer, the Internet, a web cam, a headset, and file sharing software. This new method allows for even greater ease of planning and attendance, applicability to collaboration, and mobility. You could attend a web conference on the beach while the other attendees are on a train, at home, in a restaurant, or anywhere else.
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Monday, September 14, 2009
Audio teleconference system with wireless communication
This invention is to enable transfer of a right to speak in an audio teleconference in view of usability. In this invention, a PoC server is introduced to achieve new functions, such as transfer of the right to speak, reservation to acquire the right to speak, and deprivation of the right to speak. The PoC server has a teleconference presence manager, a teleconference manager, and a teleconference audio communication manager. The right to speak is managed in a presence data storage of the teleconference presence manager. The teleconference presence manager directly manages the presence data storage, but the teleconference manager manages transfer of the right to speak. In addition, the teleconference manager manages reservation of the right to speak by a storage for a reservation list of the right to speak. The teleconference audio communication manager carries out routing of audio data. At that time, only audio data received from a user holding the right to speak is transferred.
Inventors:
Morishima, Hisayuki (Kawasaki, JP)
Suzuki, Takako (Kawasaki, JP)
Yamamoto, Yuuki (Kawasaki, JP)
Ohno, Takashi (Kawasaki, JP)
Okuyama, Satoshi (Kawasaki, JP)
Horio, Kenichi (Kawasaki, JP)
Kakuta, Jun (Kawasaki, JP)
Gunchi, Toshinao (Shinjuku, JP)
Tabuchi, Yuji (Shinjuku, JP)
Inventors:
Morishima, Hisayuki (Kawasaki, JP)
Suzuki, Takako (Kawasaki, JP)
Yamamoto, Yuuki (Kawasaki, JP)
Ohno, Takashi (Kawasaki, JP)
Okuyama, Satoshi (Kawasaki, JP)
Horio, Kenichi (Kawasaki, JP)
Kakuta, Jun (Kawasaki, JP)
Gunchi, Toshinao (Shinjuku, JP)
Tabuchi, Yuji (Shinjuku, JP)
Audio Teleconferencing
Although videoconferencing is making headway, audio teleconferencing continues to deliver on the promise of long-distance communications for education and business.
When my mother used to ask my brother questions about baseball, Bob would invariably begin his answer saying, "Mom, nine men are on a baseball team." In a similar vein, this discussion of teleconferencing hybrids must start with a short explanation of the two-wire plain old telephone service.
The standard analog telephone connection between a residential or commercial site and the local switching office is a two-conductor pair. Because both sides of the conversation travel over the two-conductor pair, some means must be provided to separate the transmit and receive signals. One two-wire to four-wire conversion is done on the switching-office side to facilitate sending to and receiving from distant switching offices. The other conversion happens at the customer's handset. The separation of the transmit signal from the receive signal is not accomplished perfectly. In particular, when the customer speaks into his handset, he hears himself in the earpiece. This is known as sidetone and gives a handset the familiar live sound. This is a classic case of "If you can't fix it, feature it."
Signal leakage reduction * The analog approach: Not more than a few years ago, if you wanted to connect a sound system to an analog phone line, you would use an analog hybrid. An analog hybrid uses passive techniques to minimize the amount of the local transmit signal that leaks through to the local receive signal. Because the impedance of the telephone line is complex and can change during a conversation, an analog hybrid can only achieve about 10 dB to 15 dB reduction in the transmit signal leakage to the receive output. This is an obvious problem in a teleconferencing application because this leakage is clearly audible from the local loudspeakers. Aside from being annoying, transmit leakage can cause reduced intelligibility or feedback in the local teleconferencing sound system. * The digital approach: With the advent of modern digital signal processors, a much more effective approach to transmit leakage reduction became feasible. Using a digital adaptive filter, the digital hybrid can realize a much better approximation to the telephone line impedance and can also track changes in the line impedance over time. Fairly typical for a digital hybrid is 30 dB to 40 dB of transmit leakage reduction. As an added bonus, once the signal is in the digital domain, other desirable signal processing can be performed by the digital signal processor.
Echo suppression There is perhaps no better known catch phrase in teleconferencing than full-duplex operation. Full duplex literally means that the transmit and receive paths are always fully open all the time. But because all digital hybrids use some form of echo suppression, none can have truly full-duplex operation. In practical terms, full-duplex means the ability for participants who are physically remote from one another to hold conversations. Whether this goal is fulfilled has to do both with the performance of the digital hybrid and with other aspects of the system implementation.
Echo suppression refers to manipulating the transmit and receive signal path gain to achieve even lower transmit-to-receive leakage. In addition, echo suppression can minimize the effects of low-level echoes caused by line reflections in the phone system.
When my mother used to ask my brother questions about baseball, Bob would invariably begin his answer saying, "Mom, nine men are on a baseball team." In a similar vein, this discussion of teleconferencing hybrids must start with a short explanation of the two-wire plain old telephone service.
The standard analog telephone connection between a residential or commercial site and the local switching office is a two-conductor pair. Because both sides of the conversation travel over the two-conductor pair, some means must be provided to separate the transmit and receive signals. One two-wire to four-wire conversion is done on the switching-office side to facilitate sending to and receiving from distant switching offices. The other conversion happens at the customer's handset. The separation of the transmit signal from the receive signal is not accomplished perfectly. In particular, when the customer speaks into his handset, he hears himself in the earpiece. This is known as sidetone and gives a handset the familiar live sound. This is a classic case of "If you can't fix it, feature it."
Signal leakage reduction * The analog approach: Not more than a few years ago, if you wanted to connect a sound system to an analog phone line, you would use an analog hybrid. An analog hybrid uses passive techniques to minimize the amount of the local transmit signal that leaks through to the local receive signal. Because the impedance of the telephone line is complex and can change during a conversation, an analog hybrid can only achieve about 10 dB to 15 dB reduction in the transmit signal leakage to the receive output. This is an obvious problem in a teleconferencing application because this leakage is clearly audible from the local loudspeakers. Aside from being annoying, transmit leakage can cause reduced intelligibility or feedback in the local teleconferencing sound system. * The digital approach: With the advent of modern digital signal processors, a much more effective approach to transmit leakage reduction became feasible. Using a digital adaptive filter, the digital hybrid can realize a much better approximation to the telephone line impedance and can also track changes in the line impedance over time. Fairly typical for a digital hybrid is 30 dB to 40 dB of transmit leakage reduction. As an added bonus, once the signal is in the digital domain, other desirable signal processing can be performed by the digital signal processor.
Echo suppression There is perhaps no better known catch phrase in teleconferencing than full-duplex operation. Full duplex literally means that the transmit and receive paths are always fully open all the time. But because all digital hybrids use some form of echo suppression, none can have truly full-duplex operation. In practical terms, full-duplex means the ability for participants who are physically remote from one another to hold conversations. Whether this goal is fulfilled has to do both with the performance of the digital hybrid and with other aspects of the system implementation.
Echo suppression refers to manipulating the transmit and receive signal path gain to achieve even lower transmit-to-receive leakage. In addition, echo suppression can minimize the effects of low-level echoes caused by line reflections in the phone system.
Equalizer
Dating as far back as the 1930's, the equalizer is the oldest and probably the most extensively used signal processing device available to the recording or sound reinforcement engineer. Today there are many types of equalizers available, and these vary greatly in sophistication, from the simple bass and treble tone control of the fifties to advanced equipment like the modern multi-band graphic equalizer and the more complex parametric types. Basically, an equalizer consists of a number of electronic filters which allow frequency response of a sound system or signal chain to be altered. Over the past half century, equalizers design has grown increasingly sophisticated. Designs began with the basic 'shelving filter', but have since evolved to meet the requirements of today's audio industry.
Understanding EQ and its Effects on Signals
There are two areas of equalization that I want to cover. Those two areas are vocals and music. I'd like to discuss the different effects of frequencies within audio signals. What do certain frequencies do for sound and how we understand those sounds. Why are some sound harsh? Why do things sound muddy? Why can't I understand the vocals? I'll try and answer all of these question and hopefully bring some light to the voodoo world of EQ.
Vocals
Roughly speaking, the speech spectrum may be divided into three main frequency bands corresponding to the speech components known as fundamentals, vowels, and consonants.
Speech fundamentals occur over a fairly limited range between about 125Hz and 250Hz. The fundamental region is important in that it allows us to tell who is speaking, and its clear transmission is therefore essential as far as voice quality is concerned.
Vowels essentially contain the maximum energy and power of the voice, occurring over the range of 350Hz to 2000Hz. Consonants occurring over the range of 1500Hz to 4000Hz contain little energy but are essential to intelligibility.
For example, the frequency range from 63 to 500Hz carries 60% of the power of the voice and yet contributes only 5% to the intelligibility. The 500Hz to 1KHz region produces 35% of the intelligibility, while the range from 1 to 8KHz produces just 5% of the power but 60% of the intelligibility.
By rolling off the low frequencies and accentuating the range from 1 to 5KHz, the intelligibility and clarity can be improved.
Here are some of the effect EQ can have in regards to intelligibility. Boosting the low frequencies from 100 to 250Hz makes a vocal boomy or chesty. A cut in the 150 to 500Hz area will make it boxy, hollow, or tube like. Dips around 500 to 1Khz produce hardness, while peaks about 1 and 3Khz produce a hard metallic nasal quality. Dips around 2 to 5KHz reduce intelligibility and make vocals woolly and lifeless. Peaks in the 4 to 10KHz produce sibilance and a gritty quality.
Important to vocal intelligibility - Too much boost between 2 and 4KHz
can mask certain vocal sounds such as 'm', 'b', 'v'. Too much boost between
1 and 4KHz can produce 'listening fatigue'. Vocals can be highlighted at the 3KHz
area and at the same time dipping the instruments at the same frequency.
Accentuation of vocals:
The range from 1.25 to 8K governs the clarity of vocals. Too much in the area of 5 to 16K can cause sibilance.
Instruments
Mic'ing instruments is an art ... and equalizers can often times be used to help an engineer get the sound he is looking for. Many instruments have complex sounds with radiating patterns that make it almost impossible to capture when close mic'ing. An equalizer can compensate for these imbalances by accenting some frequencies and rolling off others. The goal is to capture the sounds as natural as possible and use equalizers to straighten out any non-linear qualities to the tones.
Clarity of many instruments can be improved by boosting their harmonics. In fact, the ear in many cases actually fills in hard-to-hear fundamental notes of sounds, provided the harmonics are clear. Drums are one instrument that can be effectively lifted and cleaned up simply by rolling off the bass giving way to more harmonic tones.
Here are a few other pin point frequencies to start with for different instruments. In a live sound situation, I might preset the console's eq to these frequencies to help save time once the sound check is under way. These aren't the answers to everything... just a place to start at.
Kick Drum:
Besides the usual cuts in the 200Hz to 400 area, some tighter bandwidths at 380Hz, 5kHz and shelf at 10k may help. The point of these frequencies makes for space for the fundamental tones of a bass guitar or stand up. I have also found a high pass filter at 40Hz to 50Hz will help tighten up the kick along with giving your compressor a signal it can deal with musically.
Snare Drum:
The snare drum is an instrument that can really be clouded by having too much low end. Frequencies under about 150Hz are really un-usable for modern mixing styles. I would suggest a high pass filter in this case. Most snares are out front enough so a few cuts might be all that is needed. I like to start with 260Hz, boost at 5kHz, and some 12K. This are just frequencies to play with. Doesn't mean you will use all. If the snare is too transparent in the mix but I like the level it is at, a cut at 5K can give it a little more distance and that might mean a little boost at 10K to brighten it up.
High Hats:
High hats have very little low end information. I high pass at 200Hz can clean up a lot of un-usable mud in regards to mic bleed. The mid tones are the most important to a high hat. This will mean the 400Hz to 1K area but I've found the 600Hz to 800Hz area to be the most effective. To brighten up high hats, a shelving filter at 12.5K does nicely.
Toms and Floor Toms:
Again, the focus here is control. I like to high pass at 100Hz which give more focus about tone rather than size. Most toms could use a cut in the 300Hz to 400Hz area. And there is nothing real usable under 100Hz for a tom... unless you are going for a special effect. Too much low end cloud up harmonics and the natural tones of the instrument. Think color not big low end. Boost at 5kHz and shelf at 10kHz to make things more aggressive. .
Over Heads:
In my opinion, drum over heads are the most important mics on a drum kit. They are the ones that really define the sound of the drums. That also give the kit some ambience and space. These mics usually need a cut in the 400Hz area and can use a good rolling off at about 150Hz. Again, they are not used for power.... these mics 'are' the color of your drum sound. Roll off anything that will mask harmonic content or make your drums sound dull. Cuts at 800Hz can bring more focus to these mics and a little boost of a shelving filter at 12.5K can bring some air to the tones as well.
Bass Guitar:
Bass guitar puts out all the frequencies that you really don't want on every other instrument. The clarity of bass is defined a lot at 800Hz. Too much low end can mask the clarity of a bass line. I've heard other say that the best way to shape the bass tone is to roll off everything below 150Hz, mold the mids into the tone you are looking for, then slowly roll the low end back in until the power and body is there you are looking for. If the bass isn't defined enough, there is probably too much low end and not enough mid range clarity. Think of sounds in a linear fashion, like on a graph. If there is too much bass and no clarity, you would see a bump in the low end masking the top end. The use of EQ can fix those abnormalities. High pass at 40Hz to 50Hz to tighten things up. .
Guitar/piano/ etc.:
These instruments all have fundamentals in the mid range. Rolling off low end that is not needed or usable is a good idea. Even if you feel you can't really hear the low end, it still is doing something to the mix. Low end on these instruments give what I call support. The tone is in the mids. 400Hz and 800Hz are usually a point of interest as are the upper mids or 1K to 5K. Anything above that just adds brightness. Remember to look at perspective though. Is a kick brighter than a vocal? Is a piano bright than a vocal? Is a cymbal brighter than a vocal?
In Closing
Equalizers are one of the most over looked and mis-used pieces of gear in the audio industry. By understanding equalizers better, an engineer can control and get the results he or she is looking for. The key to EQ'ing is knowing how to get the results you are looking for. Also, knowing if its a mic character or mic placement problem. EQ can't fix everything. It can only change what signal its working with. Equalizers are also a lot more effective taking away things in the signal than replacing what was never there.
Understanding EQ and its Effects on Signals
There are two areas of equalization that I want to cover. Those two areas are vocals and music. I'd like to discuss the different effects of frequencies within audio signals. What do certain frequencies do for sound and how we understand those sounds. Why are some sound harsh? Why do things sound muddy? Why can't I understand the vocals? I'll try and answer all of these question and hopefully bring some light to the voodoo world of EQ.
Vocals
Roughly speaking, the speech spectrum may be divided into three main frequency bands corresponding to the speech components known as fundamentals, vowels, and consonants.
Speech fundamentals occur over a fairly limited range between about 125Hz and 250Hz. The fundamental region is important in that it allows us to tell who is speaking, and its clear transmission is therefore essential as far as voice quality is concerned.
Vowels essentially contain the maximum energy and power of the voice, occurring over the range of 350Hz to 2000Hz. Consonants occurring over the range of 1500Hz to 4000Hz contain little energy but are essential to intelligibility.
For example, the frequency range from 63 to 500Hz carries 60% of the power of the voice and yet contributes only 5% to the intelligibility. The 500Hz to 1KHz region produces 35% of the intelligibility, while the range from 1 to 8KHz produces just 5% of the power but 60% of the intelligibility.
By rolling off the low frequencies and accentuating the range from 1 to 5KHz, the intelligibility and clarity can be improved.
Here are some of the effect EQ can have in regards to intelligibility. Boosting the low frequencies from 100 to 250Hz makes a vocal boomy or chesty. A cut in the 150 to 500Hz area will make it boxy, hollow, or tube like. Dips around 500 to 1Khz produce hardness, while peaks about 1 and 3Khz produce a hard metallic nasal quality. Dips around 2 to 5KHz reduce intelligibility and make vocals woolly and lifeless. Peaks in the 4 to 10KHz produce sibilance and a gritty quality.
Important to vocal intelligibility - Too much boost between 2 and 4KHz
can mask certain vocal sounds such as 'm', 'b', 'v'. Too much boost between
1 and 4KHz can produce 'listening fatigue'. Vocals can be highlighted at the 3KHz
area and at the same time dipping the instruments at the same frequency.
Accentuation of vocals:
The range from 1.25 to 8K governs the clarity of vocals. Too much in the area of 5 to 16K can cause sibilance.
Instruments
Mic'ing instruments is an art ... and equalizers can often times be used to help an engineer get the sound he is looking for. Many instruments have complex sounds with radiating patterns that make it almost impossible to capture when close mic'ing. An equalizer can compensate for these imbalances by accenting some frequencies and rolling off others. The goal is to capture the sounds as natural as possible and use equalizers to straighten out any non-linear qualities to the tones.
Clarity of many instruments can be improved by boosting their harmonics. In fact, the ear in many cases actually fills in hard-to-hear fundamental notes of sounds, provided the harmonics are clear. Drums are one instrument that can be effectively lifted and cleaned up simply by rolling off the bass giving way to more harmonic tones.
Here are a few other pin point frequencies to start with for different instruments. In a live sound situation, I might preset the console's eq to these frequencies to help save time once the sound check is under way. These aren't the answers to everything... just a place to start at.
Kick Drum:
Besides the usual cuts in the 200Hz to 400 area, some tighter bandwidths at 380Hz, 5kHz and shelf at 10k may help. The point of these frequencies makes for space for the fundamental tones of a bass guitar or stand up. I have also found a high pass filter at 40Hz to 50Hz will help tighten up the kick along with giving your compressor a signal it can deal with musically.
Snare Drum:
The snare drum is an instrument that can really be clouded by having too much low end. Frequencies under about 150Hz are really un-usable for modern mixing styles. I would suggest a high pass filter in this case. Most snares are out front enough so a few cuts might be all that is needed. I like to start with 260Hz, boost at 5kHz, and some 12K. This are just frequencies to play with. Doesn't mean you will use all. If the snare is too transparent in the mix but I like the level it is at, a cut at 5K can give it a little more distance and that might mean a little boost at 10K to brighten it up.
High Hats:
High hats have very little low end information. I high pass at 200Hz can clean up a lot of un-usable mud in regards to mic bleed. The mid tones are the most important to a high hat. This will mean the 400Hz to 1K area but I've found the 600Hz to 800Hz area to be the most effective. To brighten up high hats, a shelving filter at 12.5K does nicely.
Toms and Floor Toms:
Again, the focus here is control. I like to high pass at 100Hz which give more focus about tone rather than size. Most toms could use a cut in the 300Hz to 400Hz area. And there is nothing real usable under 100Hz for a tom... unless you are going for a special effect. Too much low end cloud up harmonics and the natural tones of the instrument. Think color not big low end. Boost at 5kHz and shelf at 10kHz to make things more aggressive. .
Over Heads:
In my opinion, drum over heads are the most important mics on a drum kit. They are the ones that really define the sound of the drums. That also give the kit some ambience and space. These mics usually need a cut in the 400Hz area and can use a good rolling off at about 150Hz. Again, they are not used for power.... these mics 'are' the color of your drum sound. Roll off anything that will mask harmonic content or make your drums sound dull. Cuts at 800Hz can bring more focus to these mics and a little boost of a shelving filter at 12.5K can bring some air to the tones as well.
Bass Guitar:
Bass guitar puts out all the frequencies that you really don't want on every other instrument. The clarity of bass is defined a lot at 800Hz. Too much low end can mask the clarity of a bass line. I've heard other say that the best way to shape the bass tone is to roll off everything below 150Hz, mold the mids into the tone you are looking for, then slowly roll the low end back in until the power and body is there you are looking for. If the bass isn't defined enough, there is probably too much low end and not enough mid range clarity. Think of sounds in a linear fashion, like on a graph. If there is too much bass and no clarity, you would see a bump in the low end masking the top end. The use of EQ can fix those abnormalities. High pass at 40Hz to 50Hz to tighten things up. .
Guitar/piano/ etc.:
These instruments all have fundamentals in the mid range. Rolling off low end that is not needed or usable is a good idea. Even if you feel you can't really hear the low end, it still is doing something to the mix. Low end on these instruments give what I call support. The tone is in the mids. 400Hz and 800Hz are usually a point of interest as are the upper mids or 1K to 5K. Anything above that just adds brightness. Remember to look at perspective though. Is a kick brighter than a vocal? Is a piano bright than a vocal? Is a cymbal brighter than a vocal?
In Closing
Equalizers are one of the most over looked and mis-used pieces of gear in the audio industry. By understanding equalizers better, an engineer can control and get the results he or she is looking for. The key to EQ'ing is knowing how to get the results you are looking for. Also, knowing if its a mic character or mic placement problem. EQ can't fix everything. It can only change what signal its working with. Equalizers are also a lot more effective taking away things in the signal than replacing what was never there.
Audio Recording
Audio Recording
The Audio Home Recording Act of 1992 (AHRA) amended the United States copyright law by adding Chapter 10, "Digital Audio Recording Devices and Media". The act enabled the release of recordable digital formats such as Sony and Philips' Digital Audio Tape without fear of contributory infringement lawsuits.
Sound recording and reproduction is an electrical or mechanical inscription and re-creation of sound waves, such as spoken voice, singing, instrumental music, or sound effects. The two main classes of sound recording technology are analog recording and digital recording. Acoustic analog recording is achieved by a small microphone diaphragm that can detect changes in atmospheric pressure (acoustic sound waves) and record them as a graphic representation of the sound waves on a medium such as a phonograph (in which a stylus senses grooves on a record). In magnetic tape recording, the sound waves vibrate the microphone diaphragm and are converted into a varying electric current, which is then converted to a varying magnetic field by an electromagnet, which makes a representation of the sound as magnetized areas on a plastic tape with a magnetic coating on it. Analog sound reproduction is the reverse process, with a bigger loudspeaker diaphragm causing changes to atmospheric pressure to form acoustic sound waves. Electronically generated sound waves may also be recorded directly from devices such as an electric guitar pickup or a synthesizer, without the use of acoustics in the recording process other than the need for musicians to hear how well they are playing during recording sessions.
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