Note: Descriptions are shown in the official language in which they were submitted.
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AUDIO CONFERENCING SYSTEM STREAMING SUMMED CONFERENCE
SIGNAL ONTO THE INTERNET
CROSS REFERENCE TO RELATED APPLICATIONS
This application contains subject matter related to a commonly assigned co-
pending application designated serial number TBD, filed March 22, 2000,
entitled
"Scalable Audio Conference Platform ". This application is hereby incorporated
herein
by reference.
to
BACKGROUND OF THE INVENTION
The present invention relates to telephony, and in particular to an audio
conferencing platform.
Audio conferencing platforms are well known. For example, see U.S. Patents
t5 5,483,588 and 5,495,522. Audio conferencing platforms allow conference
participants
to easily schedule and conduct audio conferences with a large number of users.
In
addition, audio conference platforms are generally capable of simultaneously
supporting many conferences.
Due to the widespread popularity of the World Wide Web, Internet traffic is at
2o an all time high and rapidly increasing. In addition, the move towards IP
communications is gathering momentum. Users are currently using the Internet
as a
mechanism for retrieving streamed audio and video media streams.
There is a need for an audio conferencing system that can stream its summed
conference audio onto the Internet in real-time. This will allow a user to
listen to an
25 audio conference supported by the audio conferencing system, over the
Internet.
SUMMARY OF THE INVENTION
Briefly, according to the present invention, an audio conferencing system
comprises an audio conference mixer that receives digitized audio signals and
sums a
3o plurality of the digitized audio signals containing speech to provide a
summed
conference signal. A transcoder receives and transcodes the summed conference
signal
to provide a transcoded summed signal that is streamed onto the Internet.
In one embodiment an audio conferencing platform includes a data bus, a
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controller, and an interface circuit that receives audio signals from a
plurality of
conference participants and provides digitized audio signals in assigned time
slots over
the data bus. The audio conferencinø platform also includes a plurality of
digital signal
processors (DSPs) adapted to communicate on the TDM bus with the interface
circuit.
At least one of the DSPs sums a plurality of the digitized audio signals
associated with
conference participants who are speaking to provide a summed conference
signal. This
DSP provides the summed conference signal to at least one of the other
plurality of
DSPs, which removes the digitized audio signal associated with a speaker whose
voice
is included in the summed conference signal, thus providing a customized
conference
~o audio signal to each of the speakers.
In a preferred embodiment, the audio conferencing platform configures at least
one of the DSPs as a centralized audio mixer and at least another one of the
DSPs as an
audio processor. Significantly, the centralized audio mixer performs the step
of
summing a plurality of the digitized audio signals associated with conference
participants who are speaking, to provide the summed conference signal. The
centralized audio mixer provides the summed conference signal to the audio
processors) for post processing and routing to the conference participants.
The post
processing includes removing the audio associated with a speaker from the
conference
signal to be sent to the speaker. For example, if there are forty conference
participants
2o and three of the participants are speaking, then the summed conference
signal will
include the audio from the three speakers. The summed conference signal is
made
available on the data bus to the thirty-seven non-speaking conference
participants.
However, the three speakers each receive an audio signal that is equal to the
summed
conference signal less the digitized audio signal associated with the speaker.
Removing
2s the speaker's voice from the audio he hears reduces echoes.
The centralized audio mixer also receives DTMF detect bits indicative of the
digitized audio signals that include a DTMF tone. The DTMF detect bits may be
provided by another of the DSPs that is programmed to detect DTMF tones. If
the
digitized audio signal is associated with a speaker, but the digitized audio
signal
3o includes a DTMF tone, the centralized conference mixer will not include the
digitized
audio signal in the summed conference signal while that DTMF detect bit signal
is
active. This ensures conference participants do not hear annoying DTMF tones
in the
conference audio. When the DTMF tone is no longer present in the digitized
audio
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signal, the centralized conference mixer may include the audio signal in the
summed
conference signal.
The audio conference platform is capable of supporting a number of
simultaneous conferences (e.g., 384). As a result, the audio conference mixer
provides
a summed conference signal for each of the conferences.
Each of the digitized audio signals may be preprocessed. The preprocessing
steps include decompressing the signal (e.g., p-Law or A-Law compression), and
determining if the magnitude of the decompressed audio signal is greater than
a
detection threshold. If it is, then a speech bit associated with the digitized
audio signal
~ o is set. Otherwise, the speech bit is cleared.
These and other objects, features and advantages of the present invention will
become apparent in light of the following detailed description of preferred
embodiments thereof, as illustrated in the accompanying drawings.
t 5 BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a pictorial illustration of a conferencing system;
FIG. 2 illustrates a functional block diagram of an audio conferencing
platform
within the conferencing system of FIG. 1;
FIG. 3 is a block diagram illustration of a processor board within the audio
2o conferencing platform of FIG. 2;
FIG. 4 is a functional blo;,k diagram illustration of the resources on the
processor board of FIG. 3;
FIG. 5 is a flow chart illustration of audio processor processing for signals
received from the network interface cards over the TDM bus;
25 FIG. 6 is a flow chart illustration of the DTMF tone detection processing;
FIGs. 7A-7B together provide a flow chart illustration of the conference mixer
processing to create a summed conference signal;
FIG. 8 is a flow chart illustration of audio processor processing for signals
to be
output to the network interface cards via the TDM bus; and
3o FIG. 9 is a flow chart illustration of the transcoding performed on the
summed
conference signals) to provide "real-time" conference audio over the Internet.
DETAILED DESCRIPTION OF THE INVENTION
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FIG. 1 is a pictorial illustration of a conferencing system 20. The system 20
connects a plurality of user sites 21-23 through a switching network 24 to an
audio
conferencing platform 26. The plurality of user sites may be distributed
worldwide, or
at a company facility/campus. For example, each of the user sites 21-23 may be
in
different cities and connected to the audio platform 26 via the switching
network 24,
that may include PSTN and PBX systems. The connections between the user sites
and
the switching network 24 may include T1, E1, T3 and ISDN lines.
Each user site 21-23 preferably includes a telephone 28 and a computer/server
30. However, a conferences site may only include either the telephone or the
computer/server. The computer/server 30 may be connected via an
Internet/intranet
backbone 32 to a server 34. The audio conferencing platform 26 and the server
34 are
connected via a data link 36 (e.g., a 10/100 Baser Ethernet link). The
computer 30
allows the user to participate in a data conference simultaneous to the audio
conference
via the server 34. In addition, the user can use the computer 30 to interface
(e.g., via
a browser) with the server 34 to perform functions such as conference control,
administration (e.g., system configuration, billing, reports,...), scheduling
and account
maintenance. The telephone 28 and the computer 30 may cooperate to provide
voice
over the Internet/intranet 32 to the audio conferencing platform 26 via the
data link 36.
FIG. 2 illustrates a functional block diagram of the audio conferencing
platform
26. The audio conferencing platform 26 includes a plurality of network
interface cards
(NICs) 38-40 that receive audio information from the switching network 24
(FIG. 1).
Each NIC may be capable of handling a plurality of different trunk lines
(e.g., eight).
The data received by the NIC is generally an 8-bit p-Law or A-Law sample. The
NIC
places the sample into a memory device (not shown), which is used to output
the audio
data onto a data bus. The data bus is preferably a time division multiplex
(TDM) bus,
for example based upon the H.110 telephony standard.
The audio conferencing platform 26 also includes a plurality of processor
boards 44-46 that receive and transmit data to the NICs 38-40 over the TDM bus
42.
The NICs and the processor boards 44-46 also communicate with a controller/CPU
3o board 48 over a system bus 50. The system bus 50 is preferably based upon
the
compact PCi standard. The CPU/controller communicates with the server 34 (FIG.
1)
via the data link 36. The controller/CPU board may include a general purpose
processor such as a 200 MHz PentiumTM CPU manufactured by Intel Corporation, a
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processor from AMD or any other similar processor (including an ASIC) having
sufficient MIPS to support the present invention.
FIG. 3 is block diagram illustration of the processor- board 44 of the audio
conferencing platform. The board 44 includes a plurality of dynamically
5 programmable digital signal processors 60-65. Each digital signal processor
(DSP) is
an integrated circuit that communicates with the controller/CPU card 48 (FIG.
2) over
tire system bus 50. Specifically, the processor board 44 includes a bus
interface 68 that
interconnects the DSPs 60-65 to the system bus 50. Each DSP also includes an
associated dual port RAM (DPR) 70-75 that buffers commands and data for
to transmission between the system bus 50 and the associated DSP.
Each DSP 60-65 also transmits data over and receives data from the TDM bus
42. The processor card 44 includes a TDM bus interface 78 that performs any
necessary signal conditioning and transformation. For example, if the TDM bus
is a
H.110 bus then it includes thirty-two serial lines, as a result the TDM bus
interface
t 5 may include a serial-to-parallel and a parallel-to-serial interface. An
example, of a
serial-to-parallel and a parallel-to-serial interface is disclosed in commonly
assigned
United States Provisional Patent Application designated serial number
60/105,369 filed
October 23, 1998 and entitled "Serial-to-Parallel/Parallel-to-Serial
Conversion
Engine" . This application is hereby incorporated by reference.
2o Each DSP 60-65 also includes an associated TDM dual port RAM 80-85 that
buffers data for transmission between the TDM bus 42 and the associated DSP.
Each of the DSPs is preferably a general purpose digital signal processor IC,
such as the model number'TMS320C6201 processor available from Texas
Instruments.
The number of DSPs resident on the processor board 44 is a function of the
size of the
25 integrated circuits, their power consumption and the heat dissipation
ability of the
processor board. For example, there may be between four and ten DSPs per
processor
board.
Executable software applications may be downloaded from the controller/CPU
48 (FIG. 2) via the system bus 50 to a selected ones) of the DSPs 60-65. Each
of the
3o DSPs is also connected to an adjacent DSP via a serial data link.
FIG. 4 is a functional illustration of the DSP resources on the processor
board
44 illustrated in FIG. 3. Referring to FIGs. 3 and 4, the controller/CPU 48
(FIG. 2)
downloads executable program instructions to a DSP based upon the function
that the
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controller/CPU assigns to the DSP. For example, the controller/CPU may
download
executable program instructions for the DSP3 62 to function as an audio
conference
mixer 90, while the DSPZ 61 and the DSP4 63 may be configured as audio
processors
92, 94, respectively. DSPS 64 may be configured to perform transcoding 95 on
the
conference sums in order to provide an audio conference signal suitable for
transmission over the Internet in real-time. This feature will be discussed in
detail
hereinafter. Significantly, this allows users to listen to the audio
conference via the
Internet (i.e., using packet switched audio). Other DSPs 60, 65 may be
configured by
the controller/CPU 48 (FIG. 2) to provide services such as DTMF detection 96,
audio
to message generation 98 and music play back 100.
Each audio processor 92, 94 is capable of supporting a certain number of user
ports (i.e., conference participants). This number is based upon the
operational speed
of the various components within the processor board, and the over-all design
of the
system. Each audio processor 92, 94 receives compressed audio data 102 from
the
t 5 conference participants over the TDM bus 42.
The TDM bus 42 may support 4096 time slots, each having a bandwidth of 64
kbps. The timeslots are generally dynamically assigned by the controller/CPU
48 (FIG.
2) as needed for the conferences that are currently occurring. However, one of
ordinary skill in the art will recognize that in a static system the timeslots
may be
2o nailed up.
FIG. 5 is a flow chart illustration of processing steps 500 performed by each
audio processor on the digitized audio signals received over the TDM bus 42
from the
NICs 38-40 (FIG. 2). The executable program instructions associated with these
processing steps 500 are typically downloaded to the audio processors 92, 94
(FIG. 4)
25 by the controller/CPU 48 (FIG. 2). The download may occur during system
initialization or reconfiguration. These processing
steps 500 are executed at least once every 125 p,seconds to provide audio of
the
requisite quality.
For each of the active/assigned ports for the audio processor, step 502 reads
the
3o audio data for that port from the TDM dual port RAM associated with the
audio
processor. For example, if DSPZ 61 (FIG. 3) is configured to perform the
function of
audio processors 92 (FIG. 4), then the data is read from the read bank of the
TDM
dual port RAM 81. If the audio processor 92 is responsible for 700
active/assigned
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ports, then step 502 reads the 700 bytes of associated audio data from the TDM
dual
port RAM 81. Each audio processor includes a time slot allocation table (not
shown)
that specifies the address location in the TDM dual port RAM- for the audio
data from
each port.
Since each of the audio signals is compressed (e.g., p-Law, A-Law, etc), step
604 decompresses each of the 8-bit signals to a 16-bit word. Step 506 computes
the
average magnitude (AVM) for each of the decompressed signals associated with
the
ports assigned to the audio processor.
Step 508 is performed next to determine which of the ports are speaking. This
1 o step compares the average magnitude for the port computed in step 506
against a
predetermined magnitude value representative of speech (e.g., -35 dBm). If
average
magnitude for the port exceeds the predetermined magnitude value
representative of
speech, a speech bit associated with the port is set. Otherwise, the
associated speech
bit is cleared. Each port has an associated speech bit. Step 510 outputs all
the speech
bits (eight per timeslot) onto the TDM bus. Step 512 is performed to calculate
an
automatic gain correction (AGC) factor for each port. To compute an AGC value
for
the port, the AVM value is converted to an index value associated with a table
containing gain/attenuation factors. For example, there may be 256 index
values, each
uniquely associated with 256 gain/attenuation factors. The index value is used
by the
2o conference mixer 90 (FIG. 4) to determine the gain/attenuation factor to be
applied to
an audio signal that will be summed to create the conference sum signal.
FIG. 6 is a flow chart illustration of the DTMF tone detection processing 600.
These processing steps 600 are performed by the DTMF processor 96 (FIG. 4),
preferably at least once every 125 p.seconds, to detect DTMF tones within on
the
digitized audio signals from the NICs 38-40 (FIG. 2). One or more of the DSPs
may -
be configured to operate as a DTMF tone detector. The executable program
instructions associated with the processing steps 600 are typically downloaded
by the
controller/CPU 48 (FIG. 2) to the DSP designated to perform the DTMF tone
detection
function. The download may occur during initialization or system
reconfiguration.
3o For an assigned number of the active/assigned ports of the conferencing
system,
step 602 reads the audio data for the port from the TDM dual port RAM
associated
with the DSP(s) configured to perform the DTMF tone detection function. Step
604
then expands the 8-bit signal to a 16-bit word. Next, step 606 tests each of
these
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decompressed audio signals to determine if any of the signals includes a DTMF
tone.
For any signal that does include a DTMF tone, step 606 sets a DTMF detect bit
associated with the port. Otherwise, the DTMF detect bit is cleared. Each port
has an
associated DTMF detect bit. Step 608 informs the controller/CPU 48 (FIG. 3)
which
s DTMF tone was detected, since the tone is representative of system commands
and/or
data from a conference participant. Step 610 outputs the DTMF detect bits onto
the
TDM bus.
FIGs. 7A-7B collectively provide a flow chart illustration of processing steps
700 performed by the audio conference mixer 90 (FIG. 4) at least once every
125
to seconds to create a summed conference signal for each conference. The
executable
program instructions associated with the processing steps 700 are typically
downloaded
by the controller/CPU 48 (FIG. 2) over the system bus 50 (FIG. 2) to the DSP
designated to perform the conference mixer function. The download may occur
during
initialization or system reconfiguration.
t s Referring to FIG. 7A, for each of the active/assigned ports of the audio
conferencing system, step 702 reads the speech bit and the DTMF detect bit
received
over the TDM bus 42 (FIG. 4). Alternatively, the speech bits may be provided
over a
dedicated serial link that interconnects the audio processor and the
conference mixer.
Step 704 is then performed to determine if the speech bit for the port is set
(i.e., was
2o energy detected on that port?). If the speech bit is set, then step 706 is
performed to
see if the DTMF detect bit for the port is also set. If the DTMF detect bit is
clear,
then the audio received by the port is speech and the audio does not include
DTMF
tones. As a result, step 708 sets the conference bit for that port, otherwise
step 709
clears the conference bit associated with the port. Since the audio
conferencing
2s platform 26 (FIG. 1) can support many simultaneous conferences (e.g., 384),
the
controller/CPU 48 (FIG. 2) keeps track of the conference that each port is
assigned to
and provides that information to the DSP performing the audio conference mixer
function. Upon the completion of step 708, the conference bit for each port
has been
updated to indicate the conference participants whose voice should be included
in the
30 conference sum.
Referring to FIG. 7B, for each of the conferences, step 710 is performed to
decompress each of the audio signals associated with conference bits that are
set. Step
711 performs AGC and gain/TLP compensation on the expanded signals from step
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710. Step 712 is then performed to sum each of the compensated audio samples
to
provide a summed conference signal. Since many conference participants may be
speaking at the same time, the system preferably limits the number of
conference
participants whose voice is summed to create the conference audio. For
example, the
system may sum the audio signals from a maximum of three speaking conference
participants. Step 714 outputs the summed audio signal for the conference to
the audio
processors. In a preferred embodiment, the summed audio signal for each
conference
is output to the audio processors) over the TDM bus. Since the audio
conferencing
platform supports a number of simultaneous conferences, steps 710-714 are
performed
to for each of the conferences.
FIG. 8 is a flow chart illustration of processing steps 800 performed by each
audio processor to output audio signals over the TDM bus to conference
participants.
The executable program instructions associated with these processing steps 800
are
typically downloaded to each audio processor by the controller/CPU during
system
initialization or reconfiguration. These steps 800 are also preferably
executed at least
once every 125 p,seconds.
For each active/assigned port, step 802 retrieves the summed conference signal
for the conference that the port is assigned to. Step 804 reads the conference
bit
associated with the port, and step 806 tests the bit to determine if audio
from the port
2o was used to create the summed conference signal. If it was, then step 808
removes the
gain (e.g., AGC and gain/TLP) compensated audio signal associated with the
port from
the summed audio signal. This step removes the speaker's own voice from the
conference audio. If step 806 determines that audio from the port was not used
to
create the summed conference signal, then step 808 is bypassed. To prepare the
signal
to be output, step 810 applies a gain, and step 812 compresses the gain
corrected
signal. Step 814 then outputs the compressed signal onto the TDM bus for
routing to
the conference participant associated with the port, via the NIC (FIG. 2).
Notably, the audio conferencing platform 26 (FIG. 1) computes conference
sums at a central location. This reduces the distributed summing that would
otherwise
3o have to be performed to ensure that the ports receive the proper conference
audio. In
addition, the conference platform is readily expandable by adding additional
NICs
and/or processor boards. That is, the centralized conference mixer
architecture allows
the audio conferencing platform to be scaled to the user's requirements.
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FIG. 9 is a flow chart illustration of processing steps 900 performed by the
transcoder 95 (also referred to as an encoder). The executable program
instructions
associated with these processing steps 900 are typically downl-oaded to the
transcoding
circuit by the controller/CPU during system initialization or reconfiguration.
These
5 steps 900 are also preferably executed at least once every 125 p,seconds.
For each conference that the system is supporting - the transcoder 95 (FIG. 4)
executes step 902 to read the conference sum associated with the conference.
Step 904
is then performed to transcode the conference sum signal into a format that is
suitable
for transmission over the Internet. For example, step 904 may involve
transcoding the
10 conference sum from ~c-LAW format to a format that is suitable for
streaming the audio
conference onto the Internet in real-time. Step 906 is then performed to
output the
transcoded sum onto the system bus 50. Referring again to FIG. 2, the
transcoded sum
is output on the system bus 50 to the controller/CPU 48, which outputs the
transcoded
sum on the data link 36 to the server 34 (FIG. 1). The server then streams the
t 5 transcoded sum to conference participants via the Internet/intranet.
The transcoding may be performed using the REALPLAYERTM streamer
available from Real Networks. In general, the transcoder 95 (FIG. 4) performs
the
task of streaming audio conferences onto the Internet (and intranets) in real-
time. One
of ordinary skill in the art will recognize that transcoding/encoding
techniques other
2o than those provided by the REALPLAYERTM real-time streamer may also be
used. In
addition, the present invention is dearly not limited to the preferred
embodiment
illustrated herein. It is contemplated that the method of streaming a real-
time audio
conference to conference participants via the Internet may be performed a
number of
different ways. For example, rather than having the server physically separate
from
2s the audio conference platform, the server function may be integrated into
the audio
conference platform. In addition, the server may also receive requests/data
over the
Internet/intranet such as a question from a participant, which can be routed
to the other
conference participants either by the server in the form of text over the
Internet/intranet, a synthesized voice or the actual voice.
3o One of ordinary skill will appreciate that as processor speeds continue to
increase, that the overall system design is a function of the processing
ability of each
DSP. For example, if a sufficiently fast DSP was available, then the functions
of the
audio conference mixer, the audio processor and the DTMF tone detection and
the
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other DSP functions may be performed by a single DSP.
Although the present invention has been shown and described with respect to
several preferred embodiments thereof, various changes, omissions and
additions to the
form and detail thereof, may be made therein, without departing from the
spirit and
scope of the invention.
What is claimed is:
