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Filtering audio signals from a combined microphone/speaker earpiece    

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United States Patent5949891   
Link to this pagehttp://www.wikipatents.com/5949891.html
Inventor(s)Wagner; Larry (Oakland, CA); Wu; Carol (Beaverton, OR)
AbstractAn audio processing system applies a filter to convert the audio signals generated by the microphone of a combined microphone/speaker earpiece into filtered audio signals, where the filter is designed to correct for distortions in the audio signals that result from the microphone being part of the combined earpiece.
   














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Drawing from US Patent 5949891
Filtering audio signals from a combined microphone/speaker earpiece - US Patent 5949891 Drawing
Filtering audio signals from a combined microphone/speaker earpiece
Inventor     Wagner; Larry (Oakland, CA); Wu; Carol (Beaverton, OR)
Owner/Assignee     Intel Corporation (Santa Clara, CA)
Patent assignment
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Company News
Publication Date     September 7, 1999
Application Number     08/342,270
PAIR File History     Application Data   Transaction History
Image File Wrapper   Patent Term   Fees
Litigation
Filing Date     November 16, 1994
US Classification     381/98 381/26 381/66
Int'l Classification     H03G 005/00
Examiner     Chang; Vivian
Assistant Examiner     Nguyen; Duc
Attorney/Law Firm     Murray; William H. Kinsella; N. Stephan ,
Address
Parent Case     RELATED APPLICATIONS This application is a continuation-in-part of U.S. patent application Ser. No. 08/340,172, filed Nov. 15, 1994, entitled "Synchronizing Computer Systems" of Benjamin Vrvilo, which is a continuation-in-part of U.S. patent application Ser. No. 08/157,694, filed Nov. 24, 1993. This application is related to U.S. patent application Ser. No. 08/342,076, filed Nov. 16, 1994, entitled "Managing Bandwidth Over a Computer Network" of Robert Alexander Marshall, et al. This application is also related to U.S. patent application Ser. No. 08/305,206 (filed Sep. 13, 1994), U.S. patent application Ser. No. 08/137,319 (filed Oct. 14, 1993), U.S. patent application Ser. No. 08/170,146 (filed Dec. 20, 1993), U.S. patent application Ser. No. 08/235,955 (filed Apr. 28, 1994), and U.S. patent application Ser. No. 08/133,612 (filed Oct. 12, 1993).
Priority Data    
USPTO Field of Search     381/68.6 381/71 381/95 381/94 381/71.8 381/66 381/98 381/103 381/26 381/151 379/390 379/399 379/406 379/407 379/406 379/407
Patent Tags     filtering audio signals combined microphone/speaker earpiece
   
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 Technical Review Submit all comments and votes
 Claims Submit all comments and votes
 


What is claimed is:

1. A process for processing audio signals, comprising the steps of:

(a) generating audio signals corresponding to sounds emanating from a local user using a microphone that is part of a combined microphone/speaker earpiece;

(b) filtering the audio signals using a local audio processing system including a digital filter which is implemented as a stackable driver on a computer processor to reduce distortions in the audio signals, wherein the filter is implemented using a cascade of a second-order high-pass Chebyshev Type I Infinite Impulse Response filter and a sixth-order Infinite Impulse Response filter designed using Steiglitz approximation; and

(c) transmitting the filtered audio signals from the local audio processing system to a remote audio processing system for playback to a remote user.

2. The process of claim 1, wherein the stackable driver is implemented under a Spectron Microsystems SPOX.TM. operating system.

3. The process of claim 1, wherein the stackable driver is selectively enabled and disabled.

4. The process of claim 1, wherein the filter is designed to correct for distortions due to reverberation signals that reflect off the cheek of a user of the earpiece.

5. The process of claim 1, wherein the filter is designed to correct for distortions due to sounds that become out of phase at the microphone.

6. The process of claim 1, wherein the filter is designed to correct for distortions due to directionality/loss of relatively high frequencies of the audio signals.

7. The process of claim 1, wherein the filter produces an approximately 3 dB bump at approximately 2 kHz.

8. The process of claim 1, wherein the combined earpiece is a Plantronics Enterprise.TM. headset.

9. The process of claim 1, wherein step (b) comprises the steps of:

(1) digitizing the audio signals using the local audio processing system; and

(2) applying the digital filter to the digital audio signals using the local audio processing system to generate the filtered audio signals, and step (c) comprises the steps of:

(1) compressing the filtered audio signals using the local audio processing system; and

(2) transmitting the compressed audio signals from the local audio processing system to the remote audio processing system for decompression and playback to the remote user; and further comprising the steps of:

(d) receiving remote compressed audio signals by the local audio processing system from the remote audio processing system;

(e) decompressing the remote compressed audio signals by the local audio processing system; and

(f) playing the remote decompressed audio signals with a speaker that is part of the combined microphone/speaker earpiece.

10. The process of claim 1, wherein:

the combined earpiece is a Plantronics Enterprise.TM. headset;

the digital filter is implemented as a stackable driver on a computer processor under a Spectron Microsystems SPOX.TM. operating system;

the stackable driver is selectively enabled and disabled; and

the digital filter is implemented using a cascade of a second-order high-pass Chebyshev Type I Infinite Impulse Response filter and a sixth-order Infinite Impulse Response filter designed using Steiglitz approximation that produces an approximately 3 dB bump at approximately 2 kHz.

11. The process of claim 10, wherein the digital filter is designed to correct for distortions due to reverberation signals that reflect off the cheek of a user of the earpiece, sounds from the user's mouth that become out of phase at the microphone, and directionality/loss of relatively high frequencies of the audio signals.

12. The process of claim 10, wherein step (c) comprises the steps of:

(1) compressing the filtered audio signals using the local audio processing system; and

(2) transmitting the compressed audio signals from the local audio processing system to the remote audio processing system for decompression and playback to the remote user.

13. A system for processing audio signals, comprising:

(a) an earpiece comprising a microphone; and

(b) a local audio processing system, wherein:

the microphone is adapted to generate audio signals corresponding to sounds emanating from a local user;

the local audio processing system is adapted to apply a digital filter implemented as a stackable driver on a computer processor to the audio signals to reduce distortions in the audio signals, wherein the filter is implemented using a cascade of a second-order high-pass Chebyshev Type I Infinite Impulse Response filter and a sixth-order Infinite Impulse Response filter designed using Steiglitz approximation; and

the local audio processing system is adapted to transmit the filtered audio signals from the local audio processing system to a remote audio processing system for playback to a remote user.

14. The system of claim 13, wherein the stackable driver is implemented under a Spectron Microsystems SPOX.TM. operating system.

15. The system of claim 13, wherein the stackable driver is selectively enabled and disabled.

16. The system of clain 13, wherein the filter is designed to correct for distortions due to reverberation signals that reflect off the cheek of a user of the earpiece.

17. The system of claim 13, wherein the filter is designed to correct for distortions due to sounds that become out of phase at the microphone.

18. The system of claim 13, wherein the filter is designed to correct for distortions due to directionality/loss of relatively high frequencies of the audio signals.

19. The system of claim 13, wherein the filter is adapted to produce an approximately 3 dB bump at approximately 2 kHz.

20. The system of claim 13, wherein the combined earpiece is a Plantronics Enterprise.TM. headset.

21. The system of claim 13, wherein:

the local audio processing system is adapted to digitize the audio signals;

the local audio processing system is adapted to apply the digital filter to the digital audio signals;

the local audio processing system is adapted to compress the filtered audio signals;

the local audio processing system is adapted to transmit the compressed audio signals to the remote audio processing system for decompression and playback to the remote user;

the local audio processing system is adapted to receive remote compressed audio signals from the remote audio processing system;

the local audio processing system is adapted to decompress the remote compressed audio signals; and

a speaker of the earpiece is adapted to render the remote decompressed audio signals.

22. The system of claim 13, wherein:

the combined carpiece is a Plantronics Enterprise.TM. headset;

the digital filter is implemented as a stackable driver on a computer processor under a Spectron Microsystems SPOX.TM. operating system;

the stackable driver is selectively enabled and disabled; and

the digital filter is implemented using a cascade of a second-order high-pass Chebyshev Type I Infinite Impulse Response filter and a sixth-order Infinite Impulse Response filter designed using Steiglitz approximation that produces an approximately 3 dB bump at approximately 2 kHz.

23. The system of claim 22, wherein the digital filter is designed to correct for distortions due to reverberation signals that reflect off the cheek of a user of the earpiece, sounds from the user's mouth that become out of phase at the microphone, and directionality/loss of relatively high frequencies of the audio signals.

24. The system of claim 22, wherein:

the local audio processing system is adapted to compress the filtered audio signals; and

the local audio processing system is adapted to transmit the compressed audio signals to the remote audio processing system for decompression and playback to the remote user.

25. A process for processing audio signals, comprising the steps of:

(a) generating, with a microphone of an earpiece comprising the microphone, audio signals corresponding to sounds emanating from a local user having a head having a mouth, an ear, and a cheek between the mouth and ear;

(b) filtering, with a digital filter of a local audio processing system, the audio signals to reduce distortions in the audio signals due to the microphone being separated from the local user's mouth and in physical contact with the head when the earpiece is mounted at the local user's ear, wherein the distortions are caused by one of reverberation signals that reflect off the cheek of the local user and sounds from the local user's mouth that become out of phase at the microphone, wherein the digital filter is implemented as a stackable driver on a computer processor; and

(c) transmitting, with the local audio processing system, the filtered audio signals to a remote audio processing system for playback to a remote user.

26. A system for processing audio signals, comprising:

(a) an earpiece comprising a microphone for generating audio signals corresponding to sounds emanating from a local user having a head having a mouth, an ear, and a cheek between the mouth and ear; and

(b) a local audio processing system comprising a digital filter implemented as a stackable driver on a computer processor, wherein the local audio processing system is for filtering the audio signals with the digital filter to reduce distortions in the audio signals due to the microphone being separated from the local user's mouth and in physical contact with the head when the earpiece is mounted at the local user's ear, wherein the distortions are caused by one of reverberation signals that reflect off the cheek of the local user and sounds from the local user's mouth that become out of phase at the microphone, and the local audio processing system is for transmitting the filtered audio signals to a remote audio processing system for playback to a remote user.
 Description Submit all comments and votes
 


BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to audio/video conferencing, and, in particular, to systems for real-time audio, video, and data conferencing in windowed environments on personal computer systems.

2. Description of the Related Art

It is desirable to provide real-time audio, video, and data conferencing between personal computer (PC) systems operating in windowed environments such as those provided by versions of Microsoft.RTM. Windows.TM. operating system. There are difficulties, however, with providing real-time conferencing in non-real-time windowed environments. It is also desirable to provide conferencinig between PC systems over two or more different transports.

It is accordingly an object of this invention to overcome the disadvantages and drawbacks of the known art and to provide real-time audio, video, and data conferencing between PC systems operating in non-real-time windowed environments over two or more different transports.

It is a particular object of the present invention to provide real-time audio, video, and data conferencing between PC systems operating under a Microsoft.RTM. Windows.TM. operating system over ISDN and LAN networks.

Further objects and advantages of this invention will become apparent from the detailed description of a preferred embodiment which follows.

SUMMARY OF THE INVENTION

The present invention comprises a process and system for processing audio signals. According to a preferred embodiment, the system comprises an earpiece with a microphone, and an audio processing system, electrically connected to the microphone. The microphone generates audio signals and the audio processing system converts the audio signals into filtered audio signals by applying a filter to the audio signals. The filter is designed to correct for distortions in the audio signals that result from the microphone being part of the combined earpiece.

BRIEF DESCRIPTION OF THE DRAWINGS

Other objects, features, and advantages of the present invention will become more fully apparent from the following detailed description, the appended claims, and the accompanying drawings in which:

FIG. 1 is a block diagram representing real-time point-to-point audio, video, and data conferencing between two PC systems;

FIG. 2 is a block diagram of the hardware configuration of the conferencing system of each PC system of FIG. 1;

FIG. 3 is a block diagram of the hardware configuration of the video board of the conferencing system of FIG. 2;

FIG. 4 is a block diagram of the hardware configuration of the audio/comm (ISDN) board of the conferencing system of FIG. 2;

FIG. 5 is a block diagram of the software configuration of the conferencing system of each PC system of FIG. 1;

FIG. 6 is a block diagram of the hardware configuration of the audio/comm (ISDN) board of FIG. 4;

FIG. 7 is a block diagram of the conferencing interface layer between the conferencing applications of FIG. 5, on one side, and the comm, video, and audio managers of FIG. 5, on the other side;

FIG. 8 is a representation of the conferencing call finite state machine (FSM) for a conferencing session between a local conferencing system (i.e., caller) and a remote conferencing system (i.e., callee);

FIG. 9 is a representation of the conferencing stream FSM for each conferencing system participating in a conferencing session;

FIG. 10 is a representation of the video FSM for the local video stream and the remote video stream of a conferencing system during a conferencing session;

FIG. 11 is a block diagram of the software components of the video manager of the conferencing system of FIG. 5;

FIG. 12 is a representation of a sequence of N walking key frames;

FIG. 13 is a representation of the audio FSM for the local audio stream and the remote audio stream of a conferencing system during a conferencing session;

FIG. 14 is a block diagram of the architecture of the audio subsystem of the conferencing system of FIG. 5;

FIG. 15 is a block diagram of the interface between the audio task of FIG. 5 and the audio hardware of audio/comm (ISDN) board of FIG. 2;

FIG. 16 is a block diagram of the interface between the audio task and the comm task of FIG. 5;

FIG. 17 is a block diagram of the comm subsystem of the conferencing system of FIG. 5;

FIG. 18 is a block diagram of the comm subsystem architecture for two conferencing systems of FIG. 5 participating in a conferencing session over an ISDN connection;

FIG. 19 is a representation of the comm subsystem application FSM for a conferencing session between a local site and a remote site;

FIG. 20 is a representation of the comm subsystem connection FSM for a conferencing session between a local site and a remote site;

FIG. 21 is a representation of the comm subsystem control channel handshake FSM for a conferencing session between a local site and a remote site;

FIG. 22 is a representation of the comm subsystem channel establishment FSM for a conferencing session between a local site and a remote site;

FIG. 23 is a representation of the comm subsystem processing for a typical conferencing session between a caller and a callee;

FIG. 24 is a representation of the structure of a video packet as sent to or received from the comm subsystem of the conferencing system of FIG. 5;

FIG. 25 is a representation of the compressed video bitstream for the conferencing system of FIG. 5;

FIG. 26 is a representation of a compressed audio packet for the conferencing system of FIG. 5;

FIG. 27 is a representation of the reliable transport comm packet structure;

FIG. 28 is a representation of the unreliable transport comm packet structure;

FIG. 29 are diagrams indicating typical TII-DLM connection setup and teardown sequences;

FIGS. 30 and 31 are diagrams of the architecture of the audio/comm (ISDN) board;

FIG. 32 is a diagram of the audio/comm (ISDN) board environment;

FIG. 33 is a flow diagram of the on-demand application invocation processing of the conferencing system of FIG. 5;

FIG. 34 is a flow diagram of an example of the processing implemented within the conferencing system of FIG. 5 to manage two conferencing applications in a single conferencing session with a remote conferencing system;

FIG. 35 represents the flow of bits between two remote high-resolution counters used to maintain clock values over a conferencing network;

FIG. 36 is a flow diagram of the processing of the conferencing system of FIG. 1 to control the flow of signals over reliable channels;

FIG. 37 is a flow diagram of the preemptive priority-based transmission processing implemented by the communications subsystem of the conferencing system of FIG. 1;

FIG. 38 is a state diagram for the complete rate negotiation processing;

FIG. 39 is a state diagram for the rate negotiation processing for a called node during a 64 KBPS upgrade;

FIG. 40 is a state diagram for the rate negotiation processing for a calling node during a 64 KBPS upgrade; and

FIG. 41 is a state diagram for the rate negotiation processing in loopback mode during a 64 KBPS upgrade;

FIG. 42 is a flow diagram of the processing by the conferencing system of FIGS. 5 and 17 during the automatic transport detection implemented at install time;

FIG. 43 is a block diagram showing the network connections made by the conferencing system of FIGS. 5 and 17 during the automatic transport detection implemented at run time;

FIG. 44 is a representation of the DLMLAN packet header format;

FIG. 45 is a representation of the MDM packet header format for LAN transmissions;

FIG. 46 is a representation of the connection messages for a typical conferencing session from the perspective of the MDMs on the local and remote nodes;

FIG. 47 is a flow diagram of the video negotiation processing between two conferencing systems of FIG. 1;

FIG. 48 is a flow diagram of the call-progress processing when the placement of a conference call is successful;

FIG. 49 is a representation of the interrupt-time processing for receiving data signals by the audio/video conferencing system of FIG. 5;

FIG. 50 is a representation of the interrupt-time processing fo