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| United States Patent | 5949891 |
| Link to this page | http://www.wikipatents.com/5949891.html |
| Inventor(s) | Wagner; Larry (Oakland, CA);
Wu; Carol (Beaverton, OR) |
| Abstract | An 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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Title Information  |
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Drawing from US Patent 5949891 |
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Filtering audio signals from a combined microphone/speaker earpiece |
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| Publication Date |
September 7, 1999 |
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| Filing Date |
November 16, 1994 |
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| 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). |
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Title Information  |
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References  |
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U.S. References |
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| | Reference | Relevancy | Comments | Reference | Relevancy | Comments | 7 Tong 370/253 Jul,2006 |      Your vote accepted [0 after 0 votes] | | 5539896 Lisle
Jul,1996 |      Your vote accepted [0 after 0 votes] | | 5402497 Nishimoto 381/95 Mar,1995 |      Your vote accepted [0 after 0 votes] | | 5392223 Caci 709/218 Feb,1995 |      Your vote accepted [0 after 0 votes] | | 5390177 Nahumi 370/268 Feb,1995 |      Your vote accepted [0 after 0 votes] | | 5384772 Marshall 370/235 Jan,1995 |      Your vote accepted [0 after 0 votes] | | 5384598 Rodriguez 348/384.1 Jan,1995 |      Your vote accepted [0 after 0 votes] | | 5374952 Flohr
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Dec,1991 |      Your vote accepted [0 after 0 votes] | | 5072442 Todd
Dec,1991 |      Your vote accepted [0 after 0 votes] | | 5056136 Smith 380/211 Oct,1991 |      Your vote accepted [0 after 0 votes] | | 5014267 Tompkins 370/259 May,1991 |      Your vote accepted [0 after 0 votes] | | 4998243 Kao 370/264 Mar,1991 |      Your vote accepted [0 after 0 votes] | | 4888795 Ando 348/14.14 Dec,1989 |      Your vote accepted [0 after 0 votes] | | 4731850 Levitt 381/317 Mar,1988 |      Your vote accepted [0 after 0 votes] | | 4475193 Brown 370/295 Oct,1984 |      Your vote accepted [0 after 0 votes] | | 3786188 Allen 704/263 Jan,1974 |      Your vote accepted [0 after 0 votes] | | |
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Market Review  |
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Technical Review  |
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Claims  |
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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. |
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Claims  |
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Description  |
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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 | | |