Patent Application
20110103577
This invention concerns microphones for use in computer systems
The invention has been created without the sponsorship or funding of any federally sponsored research or development program.
Not applicable.
There are many types of microphones available in the market today including interfaces that support direct wiring to an analog and digital input channels, microphones that include digital to analog converters, USB serial microphones, wireless USB microphones, directional, and omni-directional, wearable microphones, and desktop microphones to name a few of the features. As is well known, microphones are used in a variety of devices including telephones, headsets, radio transmitters, embedded into computers and other devices such as GPS and plain desk top microphones.
While there appears to be endless types of microphones available in the present market, one area where a specific microphone is needed but there is not one available is a directional microphone that is wired or wireless that supports network protocols including TCP/IP, UDP, SIP and VoIP. Without the networking feature, traditional audio input channels become cumbersome when scaling to support high numbers of users. There are SIP conference telephones however these telephones do not meet the needs because they do not connect a specific user to a specific telephone line or channel which is advantageous when separating speech input from multiple users. Nor do typical conference telephones contain features to reduce sound to a specific audio input cone located in front of the microphone as is needed to separate out people in a conference room environment. Lacking these features has disadvantages if used in a system as described in U.S. Pat. Nos. 7,047,192, and 7,603,273 or event recording systems. A directional microphone that that also has features typically found with a telephone would be more suited for such an event recording system application.
Some of the problems encountered when creating this type of device includes:
This SIP based Voice Over IP microphone would be used in a system as described by U.S. Pat. Nos. 7,047,192, and 7,603,273 or event recording systems in this way. Instead of using traditional audio equipment channels with traditional microphones, this system is based off using a telephone PBX. The advantages of using a PBX are scalability and cost reduction when compared to the traditional audio conference recording systems. Another advantage is that control of the system could be done centrally or in a distributed means by each user.
Since each microphone has ATA functionality, every microphone would be registered to the telephone PBX system, dial into the telephone conference using Session Initiated Protocol either automatically or manually.
Registration and call in could be done automatically using initialization files, web page parameters that have features to automatically connect in by dialing an extension, or a telephone keypad at each microphone. Another option may be a single button that is pressed to join the conference whereas pressing the button causes the microphone electronics and software to automatically dial the conference number similar to a memory dial on a standard telephone.
To operate the system, a typical telephone conference call would be created by a moderator and each person in the conference room would have the ability to be added to the recording.
Once all attendees have joined, participated in the conference, and the conference has ended, the recording function stops and the audio files are available for reviewing. Optionally automatic speech recognition could be added to have the conference transcribed and a small display on the microphone could present the text as it is being transcribed. The display may be a simple terminal output display with the text data being passed to the terminal via the Ethernet and TCP/IP protocol.
There could also be audio output at each microphone passed through the SIP audio channel to the ATA to the analog telephone logic with output to the speaker. The audio could be delivered to each person via a headset. The headset could also be used for language translation during the conference. Alternatively there could be a small output speaker at each microphone providing audio to the conference room from remote attendees. Audio from microphones inside the conference room would be filter out from audio output at each microphone in the room. This would avoid negative affects such as audio feedback and delayed audio from attendees speaking in the room. Filtering out audio could be accomplished by intercepting and removing network packets from specific TCP/IP addresses or specific port numbers being used as audio output for example. Another option would be to have no output speakers on each microphone but instead to have an output speaker separate from the microphones located in the conference room where attendees are located.
The microphones could be connected by a network medium like category 5 network wire. In this configuration power over Ethernet switches could be used to supply power to the microphone and related electronics. Alternatively the microphones could be a wireless network configuration using batteries for power. This configuration may be more desirable if setting up to record a conference at a single event single location and then to move to recording another event at a different time and location.
An Event Recording System as described by Poirier in U.S. Pat. Nos. 7,047,192, and 7,603,273 could be setup on a mobile PC (Laptop) with a wireless router or access point and a software PBX as are well know in the VOIP industry. The microphones would then connect via the wireless access point to the PBX which the event recording system is also connected to. Alternatively the microphones could connect directly to the event recording system using SIP or a peer to peer model like Skype or General Voice's Kontext peer to peer event recording system.
Power supply, power management or converters, an audio preamp that adjusts microphone voltage levels to telephone handset circuit levels, control features like a telephone keypad, software initialization files, an enclosure that meets physical aspects for form factor and weighting, and microphone with sufficient audio quality, and optionally a display output with a display driver circuit and a microprocessor capable of connecting Ethernet with TCP/IP and related software to present the text on the display.
Pre-amplification is achieved by incorporating an audio preamp between the microphone and the analog telephone receiver where the handset microphone is typically connected. The preamp also provides circuit balancing as required by some types of microphones. The audio preamp should have a gain control potentiometer to enable adjustments to be made to acquire the best audio input results.
This is achieved by adding a battery as the power source for the electric microphone. Alternatively power from the ATA power supply may be used but in most cases the voltage and/or current requirements do not match the microphone requirements so an additional voltage conversion and regulation circuit may be required. DC to DC converters are well known and will not be described here. If using “Power over Ethernet” supplied from a network switch maximum power rating per switch and compatible voltages may again be require for both the ATA and the microphone. The same is true to power the optional microprocessor with the display.
The goal is to take audio input from a narrow directional cone that filters out sound that's not directly in front of the microphone. Unidirectional ability provides the feature of allowing people to be sitting in a room next to each other and to have the microphone pick up audio only from the person in front of the microphone, There are different methods to achieved this affect, one is by adding a focusing tube over the end of a unidirectional microphone creating a narrow tunnel where sound can enter. Another option would be to adjust the audio gain on a unidirectional microphone to a minimum working level. A combination of the tube and the audio gain adjustment could also be used.
The typical desktop microphone form factor would allow users to have familiarity with such a device and potentially not require special training to operate the device.
Moreover, if the microphone has a standard telephone keypad with traditional buttons including mute this would also reduce the training need.
Specific features of the microphone could include:
Referring to
This application claims the benefit under 35 USC 119(e) of U.S. Provisional Patent Application No. 61/257,161, filed Nov. 2, 2009, all of which is hereby incorporated by reference.
| Number | Date | Country | |
|---|---|---|---|
| 61257161 | Nov 2009 | US |
