PATENT DOCUMENT

Publication Number: US-8456508-B2
Application Number: US-95590210-A
Country: US
Kind Code: B2

Title: Audio processing in a multi-participant conference

Abstract:
Some embodiments provide an architecture for establishing multi-participant audio conferences over a computer network. This architecture has a central distributor that receives audio signals from one or more participants. The central distributor mixes the received signals and transmits them back to participants. In some embodiments, the central distributor eliminates echo by removing each participant&#39;s audio signal from the mixed signal that the central distributor sends to the particular participant.

Claims:
What claimed is: 
     
       1. A non-transitory computer readable medium of a device of a first participant of a multi-participant conference, the computer readable medium storing a computer program which when executed by at least one processor of the device distributes audio content in a multi-participant conference, the computer program comprising sets of instructions to:
 receive audio signals from at least second and third participants of the conference; 
 generate data representative of strengths of the received audio signals; 
 generate mixed audio signals from the received audio signals; 
 to each mixed audio signal, appending a set of the generated strength data for the audio signals that are mixed to produce each mixed audio signal; and 
 transmit the mixed audio signals with the strength data to the second and third participants of the conference. 
 
     
     
       2. The non-transitory computer readable medium of  claim 1 , wherein the computer program further comprises a set of instructions to determine a strength of each received audio signal, wherein the set of instructions to determine the strength comprise a set of instructions to calculate the strength of each received audio signal as a root mean square (RMS) power of the received audio signal. 
     
     
       3. The non-transitory computer readable medium of  claim 1 , wherein the set of instructions to generate a mixed audio signal for a participant of the multi-participant conference comprises a set of instructions to remove the audio signal of the participant. 
     
     
       4. The non-transitory computer readable medium of  claim 1 , wherein the appended strength data is configured to be used by devices of the second and third participants to pan the mixed audio signal across audio loudspeakers. 
     
     
       5. The non-transitory computer readable medium of  claim 1 , wherein the set of instructions to generate the mixed audio signals comprises a set of instructions to generate a first mixed audio signal for the second participant and generating a second, different mixed audio signal for the third participant. 
     
     
       6. The non-transitory computer readable medium of  claim 1 , wherein the mixed audio signals are transmitted using real time protocol (RTP) packets comprising the strength data for each of the audio signals in the mixed audio signals. 
     
     
       7. The non-transitory computer readable medium of  claim 1 , wherein the computer program comprises a set of instructions to locally output the mixed audio signals at the device of the first participant. 
     
     
       8. A method of distributing audio content in a multi-participant conference, the method comprising:
 at a device of a first participant of the conference:
 receiving audio signals from at least second and third participants of the conference; 
 generating a single mixed audio signal comprising an audio signal locally captured by the device of the first participant and the received audio signals; 
 transmitting the mixed audio signal to the second and third participants; and 
 removing the audio signal of the device of the first participant from the mixed audio signal in order to locally output the mixed audio signal with audio signal locally captured by the device of the first participant removed. 
 
 
     
     
       9. The method of  claim 8  comprising generating strength data that represents strengths of (i) the audio signal captured by the device of the first participant and (ii) each of the received audio signals. 
     
     
       10. The method of  claim 9 , wherein the strength data comprises the strength of each of the audio signals as a root mean square (RMS) of the audio signals. 
     
     
       11. The method of  claim 8 , wherein the single mixed audio signal is generated by the device of the first participant using the audio signals of the first, second, and third participants. 
     
     
       12. The method of  claim 8  further comprising locally outputting the mixed audio signal without the audio signal captured by the device of the first participant. 
     
     
       13. The method of  claim 9 , wherein the strength data is for panning the mixed audio signal across audio loudspeakers of the device of the first participant and devices associated with the second and third participants. 
     
     
       14. The method of  claim 8 , wherein the device generates, during the conference, the mixed audio signal, a second mixed audio signal to be provided to the second participant with the received audio signal of the second participant removed, a third mixed audio signal to be provided to the third participant with the received audio signal of the third participant removed, wherein devices associated with the second and third participants do not generate any mixed audio signals during the conference. 
     
     
       15. A non-transitory computer readable medium of a first device of a first participant of a multi-participant conference, the computer readable medium storing a computer program which when executed by at least one processor of the device receives audio content in a multi-participant conference, the computer program comprising sets of instructions to:
 receive a mixed audio signal from a second device of a second participant of the conference, the mixed audio signal comprising (i) an audio signal sent from the first device captured by the first device and sent to the second device, (ii) audio captured by the second device, and (iii) an audio signal captured by a third device and sent to the second device; 
 remove the audio signal of the first device from the mixed audio signal; and 
 output the mixed audio signal at the first device with the audio signal of the first device removed. 
 
     
     
       16. The non-transitory computer readable medium of  claim 15 , wherein the computer program comprises a set of instructions to send the audio signal of the first device to the second device of the second participant. 
     
     
       17. The non-transitory computer readable medium of  claim 15 , wherein the computer program comprises a set of instructions to receive strength data that represents strengths of audio signals received at the second device of the second participant. 
     
     
       18. The non-transitory computer readable medium of  claim 17 , wherein the set of instructions to output the mixed audio signal comprises a set of instructions to use the strength data is to pan the mixed audio signal across audio loudspeakers of the first device. 
     
     
       19. The non-transitory computer readable medium of  claim 17 , wherein the strength data is generated by the second device by calculating a strength of each audio signal of the mixed audio signal as a root mean square (RMS) power of the audio signal. 
     
     
       20. The non-transitory computer readable medium of  claim 15 , wherein the second device generates mixed audio signals during the conference and the first and third devices do not generate mixed audio signals during the conference.

Description:
CLAIM OF BENEFIT TO PRIOR APPLICATIONS 
     This application is a continuation application of U.S. patent application Ser. No. 11/118,555, entitled “Audio Processing in a Multi-Participant Conference,” filed Apr. 28, 2005, now published as U.S. Publication 2006/0247045. U.S. Publication 2006/0247045 is incorporated herein by reference. 
    
    
     FIELD OF THE INVENTION 
     The present invention relates to audio processing in a multi-participant conference. 
     BACKGROUND OF THE INVENTION 
     With proliferation of general-purpose computers, there has been an increase in demand for performing conferencing through personal or business computers. In such conferences, it is desirable to identify quickly the participants that are speaking at any given time. Such identification, however, becomes difficult as more participants are added, especially for participants that only receive audio data. This is because prior conferencing applications do not provide any visual or auditory cues to help identify active speakers during a conference. Therefore, there is a need in the art for conferencing applications that assist a participant in quickly identifying the active speaking participants of the conference. 
     SUMMARY OF THE INVENTION 
     Some embodiments provide an architecture for establishing multi-participant audio conferences over a computer network. This architecture has a central distributor that receives audio signals from one or more participants. The central distributor mixes the received signals and transmits them back to participants. In some embodiments, the central distributor eliminates echo by removing each participant&#39;s audio signal from the mixed signal that the central distributor sends to the particular participant. 
     In some embodiments, the central distributor calculates a signal strength indicator for every participant&#39;s audio signal and passes the calculated indicia along with the mixed audio signal to each participant. Some embodiments then use the signal strength indicia to display audio level meters that indicate the volume levels of the different participants. In some embodiments, the audio level meters are displayed next to each participant&#39;s picture or icon. Some embodiments use the signal strength indicia to enable audio panning. 
     In some embodiments, the central distributor produces a single mixed signal that includes every participant&#39;s audio. This stream (along with signal strength indicia) is sent to every participant. When playing this stream, a participant will mute playback if that same participant is the primary contributor. This scheme provides echo suppression without requiring separate, distinct streams for each participant. This scheme requires less computation from the central distributor. Also, through IP multicasting, the central distributor can reduce its bandwidth requirements. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The novel features of the invention are set forth in the appended claims. However, for purpose of explanation, several embodiments are set forth in the following figures. 
         FIG. 1  illustrates an example of the audio/video conference architecture of some embodiments of the invention. 
         FIGS. 2 and 3  illustrate how some embodiments exchange audio content in a multi-participant audio/video conference. 
         FIG. 4  shows the software components of the audio/video conferencing application of some embodiments of the invention. 
         FIG. 5  illustrates the focus point module of some embodiments of the invention. 
         FIG. 6  is a flow chart showing mixed audio generation by the focus point in some of the embodiments. 
         FIG. 7  illustrates how the RTP protocol is used by the focus point module in some embodiments to transmit audio content. 
         FIG. 8  illustrates the non-focus point of some embodiments of the invention. 
         FIG. 9  illustrates how the RTP protocol is used by the non-focus point module in some embodiments to transmit audio content 
         FIG. 10  conceptually illustrates the flow of non-focus point decoding operation in some embodiments. 
         FIG. 11  illustrates the audio level meters displayed on some embodiments of the invention. 
         FIG. 12  shows an exemplary arrangement of participants&#39; images on one of the participants&#39; display. 
         FIG. 13  is a flow chart illustrating the process by which some embodiments of the invention perform audio panning. 
     
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
     In the following description, numerous details are set forth for the purpose of explanation. However, one of ordinary skill in the art will realize that the invention may be practiced without the use of these specific details. In other instances, well-known structures and devices are shown in block diagram form in order not to obscure the description of the invention with unnecessary detail. 
     Some embodiments provide an architecture for establishing multi-participant audio/video conferences. This architecture has a central distributor that receives audio signals from one or more participants. The central distributor mixes the received signals and transmits them back to participants. In some embodiments, the central distributor eliminates echo by removing each participant&#39;s audio signal from the mixed signal that the central distributor sends to the particular participant. 
     In some embodiments, the central distributor calculates a signal strength indicator for every participant&#39;s audio signal and passes the calculated indicia along with the mixed audio signal to each participant. Some embodiments then use the signal strength indicia to display audio level meters that indicate the volume levels of the different participants. In some embodiments, the audio level meters are displayed next to each participant&#39;s picture or icon. Some embodiments use the signal strength indicia to enable audio panning. 
     Several detailed embodiments of the invention are described below. In these embodiments, the central distributor is the computer of one of the participants of the audio/video conference. One of ordinary skill will realize that other embodiments are implemented differently. For instance, the central distributor in some embodiments is not the computer of any of the participants of the conference. 
     I. Overview 
       FIG. 1  illustrates an example of conference architecture  100  of some embodiments of the invention. This architecture allows multiple participants to engage in a conference through several computers that are connected by a computer network. In the example illustrated in  FIG. 1 , four participants A, B, C, and D are engaged in the conference through their four computers  105 - 120  and a network (not shown) that connects these computers. The network that connects these computers can be any network, such as a local area network, a wide area network, a network of networks (e.g., the Internet), etc. 
     The conference can be an audio/video conference, or an audio only conference, or an audio/video conference for some participants and an audio only conference for other participants. During the conference, the computer  105  of one of the participants (participant D in this example) serves as a central distributor of audio and/or video content (i.e., audio/video content), as shown in  FIG. 1 . This central distributor  125  will be referred to below as the focus point of the multi-participant conference. The computers of the other participants will be referred to below as non-focus machines or non-focus computers. 
     Also, the discussion below focuses on the audio operations of the focus and non-focus computers. The video operation of these computers is further described in U.S. patent application entitled “Video Processing in a Multi-Participant Video Conference”, filed concurrently with this application, with the attorney docket number APLE.P0091. In addition, U.S. patent application entitled “Multi-Participant Conference Setup”, filed concurrently with this application, with the attorney docket number APLE.P0084, describes how some embodiments set up a multi-participant conference through a focus-point architecture, such as the one illustrated in  FIG. 1 . Both these applications are incorporated herein by reference. 
     As the central distributor of audio/video content, the focus point  125  receives audio signals from each participant, mixes and encodes these signals, and then transmits the mixed signal to each of the non-focus machines.  FIG. 2  shows an example of such audio signal exchange for the four participant example of  FIG. 1 . Specifically,  FIG. 2  illustrates the focus point  125  receiving compressed audio signals  205 - 215  from other participants. From the received audio signals  205 - 215 , the focus point  125  generates a mixed audio signal  220  that includes each of the received audio signals and the audio signal from the participant using the focus point computer. The focus point  125  then compresses and transmits the mixed audio signal  220  to each non-focus machine  110 ,  115 , and  120 . 
     In the example illustrated in  FIG. 2 , the mixed audio signal  220  that is transmitted to each particular non-focus participant also includes the audio signal of the particular non-focus participant. In some embodiments, however, the focus point removes a particular non-focus participant&#39;s audio signal from the mixed audio signal that the focus point transmits to the particular non-focus participant. In these embodiments, the focus point  125  removes each participant&#39;s own audio signal from its corresponding mixed audio signal in order to eliminate echo when the mixed audio is played on the participant computer&#39;s loudspeakers. 
       FIG. 3  illustrates an example of this removal for the example illustrated in  FIG. 2 . Specifically,  FIG. 3  illustrates (1) for participant A, a mixed audio signal  305  that does not have participant A&#39;s own audio signal  205 , (2) for participant B, a mixed audio signal  310  that does not have participant B&#39;s own audio signal  210 , and (3) for participant C, a mixed audio signal  315  that does not have participant C&#39;s own audio signal  215 . 
     As shown in  FIG. 3 , the focus point  125  in some embodiments calculates signal strength indicia for the participants&#39; audio signals, and appends the signal strength indicia to the mixed signals that it sends to each participant. The non-focus computers then use the appended signal strength indicia to display audio level meters that indicate the volume levels of the different participants. In some embodiments, the audio level meters are displayed next to each participant&#39;s picture or icon. 
     Some embodiments also use the transmitted signal strength indicia to pan the audio across the loudspeakers of a participant&#39;s computer, in order to help identify orators during the conference. This panning creates an effect such that the audio associated with a particular participant is perceived to originate from a direction that reflects the on-screen position of that participant&#39;s image or icon. The panning effect is created by introducing small delays to the left or right channels. The positional effect relies on the brain&#39;s perception of small delays and phase differences. Audio level meters and audio panning are further described below. 
     Some embodiments are implemented by an audio/video conference application that can perform both focus and non-focus point operations. For example, in some embodiments, a computer readable medium may store a computer program for distributing audio content in a multi-participant audio/video conference having a central distributor of audio content. The program may include instructions. The instructions may include receiving audio signals from each participant, generating mixed audio signals from the received audio signals, and transmitting the audio signals to each participant.  FIG. 4  illustrates a software architecture for one such application. Specifically, this figure illustrates an audio/video conference application  405  that has two modules, a focus point module  410  and a non-focus point module  415 . Both these modules  410  and  415 , and the audio/video conference application  405 , run on top an operating system  420  of a conference participant&#39;s computer. 
     During a multi-participant conference, the audio/video conference application  405  uses the focus point module  410  when this application is serving as the focus point of the conference, or uses the non-focus point module  415  when it is not serving as the focus point. The focus point module  410  performs focus point audio-processing operations when the audio/video conference application  405  is the focus point of a multi-participant audio/video conference. On the other hand, the non-focus point module  415  performs non-focus point, audio-processing operations when the application  405  is not the focus point of the conference. In some embodiments, the focus and non-focus point modules  410  and  415  share certain resources. 
     The focus point module  410  is described in Section II of this document, while the non-focus point module  415  is described in Section III. 
     II. The Focus Point Module 
       FIG. 5  illustrates the focus point module  410  of some embodiments of the invention. The focus point module  410  is shown during an audio/video conferencing with multiple participants. In order to generalize the focus point operations, the example in  FIG. 5  is illustrated as having an arbitrary number of participants. This arbitrary number is denoted as “n”, which represents a number greater than 2. The focus point module  410  generates mixed audio signals for transmitting to non-focus participants, and performs audio presentation for the conference participant who is using the focus point computer during the video conference. For its audio mixing operation, the focus point module  410  utilizes (1) one decoder  525  and one intermediate buffer  530  for each incoming audio signal, (2) an intermediate buffer  532  for the focus point audio signal, (3) one audio capture module  515 , (3) one audio signal strength calculator  580 , and (4) one audio mixer  535  for each transmitted mixed audio signal, and one encoder  550  for each transmitted mixed audio signal. For its audio presentation operation at the focus-point computer, the focus point module  410  also utilizes one audio mixer  545 , one audio panning control  560  and one level meter control  570 . 
     The audio mixing operation of the focus point module  410  will now be described by reference to the mixing process  600  that conceptually illustrates the flow of operation in  FIG. 6 . The audio presentation operation of the focus point module is described in Section III below, along with the non-focus point module&#39;s audio presentation. 
     During the audio mixing process  600 , two or more decoders  525  receive (at  605 ) two or more audio signals  510  containing digital audio samples from two or more non-focus point modules. In some embodiments, the received audio signals are encoded by the same or different audio codecs at the non-focus computers. Examples of such codecs include Qualcomm PureVoice, GSM, G.711, and ILBC audio codecs. 
     The decoders  525  decode and store (at  605 ) the decoded audio signals in two or more intermediate buffers  530 . In some embodiments, the decoder  525  for each non-focus computer&#39;s audio stream uses a decoding algorithm that is appropriate for the audio codec used by the non-focus computer. This decoder is specified during the process that sets up the audio/video conference. 
     The focus point module  410  also captures audio from the participant that is using the focus point computer, through microphone  520  and the audio capture module  515 . Accordingly, after  605 , the focus point module (at  610 ) captures an audio signal from the focus-point participant and stores this captured audio signal in its corresponding intermediate buffer  532 . 
     Next, at  615 , the audio signal strength calculator  580  calculates signal strength indicia corresponding to the strength of each received signal. Audio signal strength calculator  580  assigns a weight to each signal. In some embodiments, the audio signal strength calculator  580  calculates the signal strength indicia as the Root Mean Square (RMS) power of the audio stream coming from the participant to the focus point. The RMS power is calculated from the following formula: 
               RMS   =           ∑     i   =   1     N     ⁢       (     Sample   i     )     2       N         ,         
where N is the number of samples used to calculate the RMS power and Sample, is the i th  sample&#39;s amplitude. The number of samples, N, that audio signal strength calculator  580  uses to calculate RMS value depends on the sampling rate of the signal. For example, in some embodiments of the invention where the sampling rate is 8 KHz, the RMS power might be calculated using a 20 ms chunk of audio data containing 160 samples. Other sampling rates may require a different number of samples.
 
     Next, at  620 , process  600  utilizes the audio mixers  535  and  545  to mix the buffered audio signals. Each audio mixer  535  and  545  generates mixed audio signals for one of the participants. The mixed audio signal for each particular participant includes all participants&#39; audio signals except the particular participant&#39;s audio signal. Eliminating a particular participant&#39;s audio signal from the mix that the particular participant receives eliminates echo when the mixed audio is played on the participant computer&#39;s loudspeakers. The mixers  535  and  545  mix the audio signals by generating (at  620 ) a weighted sum of these signals. To obtain an audio sample value at a particular sample time in a mixed audio signal, all samples at the particular sampling time are added based on the weight values computed by the audio signal strength calculator  580 . In some embodiments, the weights are dynamically determined based on signal strength indicia calculated at  615  to achieve certain objectives. Example of such objectives include (1) the elimination of weaker signals, which are typically attributable to noise, and (2) the prevention of one participant&#39;s audio signal from overpowering other participants&#39; signals, which often results when one participant consistently speaks louder than the other or has better audio equipment than the other. 
     In some embodiments, the mixers  535  and  545  append (at  625 ) the signal strength indicia of all audio signals that were summed up to generate the mixed signal. For instance,  FIG. 7  illustrates an RTP (Real-time Transport Protocol) audio packet  700  that some embodiments use to send a mixed audio signal  705  to a particular participant. As shown in this figure, signal strength indicia  710 - 720  are attached to the end of the RTP packet  705 . 
     Next, for the non-focus computers&#39; audio, the encoders  550  (at  630 ) encode the mixed audio signals and send them (at  635 ) to their corresponding non-focus modules. The mixed audio signal for the focus point computer is sent (at  635 ) unencoded to focus point audio panning control  560 . Also, at  635 , the signal strength indicia is sent to the level meter  570  of the focus point module, which then generates the appropriate volume level indicators for display on the display device  575  of the focus point computer. 
     After  635 , the audio mixing process  600  determines (at  640 ) whether the multi-participant audio/video conference has terminated. If so, the process  600  terminates. Otherwise, the process returns to  605  to receive and decode incoming audio signals. 
     One of ordinary skill will realize that other embodiments might implement the focus point module  410  differently. For instance, in some embodiments, the focus point  410  produces a single mixed signal that includes every participant&#39;s audio. This stream along with signal strength indicia is sent to every participant. When playing this stream, a participant will mute playback if that same participant is the primary contributor. This scheme saves focus point computing time and provides echo suppression without requiring separate, distinct streams for each participant. Also, during IP multicast, the focus point stream bandwidth can be reduced. In these embodiments, the focus point  410  has one audio mixer  535  and one encoder  550 . 
     III. The Non-Focus Point Module 
       FIG. 8  illustrates a non-focus point module  415  of an audio/video conference of some embodiments of the invention. In this example, the non-focus point module  415  utilizes a decoder  805 , two intermediate buffers  810  and  880 , a level meter control  820 , an audio panning control  845 , an audio capture module  875 , and an encoder  870 . 
     The non-focus point module performs encoding and decoding operations. During the encoding operation, the audio signal of the non-focus point participant&#39;s microphone  860  is captured by audio capture module  875  and is stored in its corresponding intermediate buffer  880 . The encoder  870  then encodes the contents of the intermediate buffer  880  and sends it to the focus point module  410 . 
     In some embodiments that use Real-time Transport Protocol (RTP) to exchange audio signals, the non-focus participant&#39;s encoded audio signal is sent to the focus point module in a packet  900  that includes RTP headers  910  plus encoded audio  920 , as shown in  FIG. 9 . 
     The decoding operation of the non-focus point module  415  will now be described by reference to the process  1000  that conceptually illustrates the flow of operation in  FIG. 10 . During the decoding operation, the decoder  805  receives (at  1005 ) audio packets from the focus point module  410 . The decoder  805  decodes (at  1010 ) each received audio packet to obtain mixed audio data and the signal strength indicia associated with the audio data. The decoder  805  saves (at  1010 ) the results in the intermediate buffer  810 . 
     The signal strength indicia are sent to level meter control  820  to display (at  1015 ) the audio level meters on the non-focus participant&#39;s display  830 . In a multi-participant audio/video conference, it is desirable to identify active speakers. One novel feature of the current invention is to represent the audio strengths by displaying audio level meters corresponding to each speaker&#39;s voice strength. Level meters displayed on each participant&#39;s screen express the volume level of the different participants while the mixed audio signal is being heard from the loud speakers  855 . Each participant&#39;s volume level can be represented by a separate level meter, thereby, allowing the viewer to know the active speakers and the audio level from each participant at any time. 
     The level meters are particularly useful when some participants only receive audio signals during the conference (i.e., some participants are “audio only participants”). Such participants do not have video images to help provide a visual indication of the participants that are speaking.  FIG. 11  illustrates an example of the use of level meters in an audio only conference of some embodiments. In this figure, each participant&#39;s audio level  1110 - 1115  is placed next to that participant&#39;s icon  1120 - 1125 . As illustrated in  FIG. 11 , some embodiments display the local microphone&#39;s voice level  1130  separately at the bottom of the screen. One of ordinary skill in the art should realize that  FIG. 11  is just one example of the way to show the level meters on a participant&#39;s display. Other display arrangements can be made without deviating from the teachings of this invention for calculating and displaying the relative strength of audio signals in a conference. 
     After  1015 , the decoded mixed audio signal and signal strength indicia stored in the intermediate buffer  810  are sent (at  1020 ) to the audio panning control  845  to control the non-focus participant&#39;s loudspeakers  855 . The audio panning operation will be further described below by reference to  FIGS. 12 and 13 . 
     After  1020 , the audio decoding process  1000  determines (at  1025 ) whether the multi-participant audio/video conference has terminated. If so, the process  1000  terminates. Otherwise, the process returns to  1005  to receive and decode incoming audio signals. 
     The use of audio panning to make the perceived audio location match the video location is another novel feature of the current invention. In order to illustrate how audio panning is performed,  FIG. 12  illustrates an example of a video-conference display presentation  1200  in the case of four participants in a video conference. As shown in  FIG. 12 , the other three participants&#39; images  1205 - 1215  are displayed horizontally in the display presentation  1200 . The local participant&#39;s own image  1220  is optionally displayed with a smaller size relative to the other participants&#39; images  1205 - 1215  at the bottom of the display presentation  1200 . 
     Some embodiments achieve audio panning through a combination of signal delay and signal amplitude adjustment. For instance, when the participant whose image  1205  is placed on the left side of the screen speaks, the audio coming from the right speaker is changed by a combination of introducing a delay and adjusting the amplitude to make the feeling that the voice is coming from the left speaker. 
       FIG. 13  illustrates a process  1300  by which the audio panning control of the non-focus module  845  operate in some embodiments of the invention. The signal strength indicia of each audio signal in the mixed audio signal is used (at  1310 ) to identify the most-contributing participant in the decoded mixed audio signal. Next, the process identifies (at  1315 ) the location of the participant or participants identified at  1310 . The process then uses (at  1320 - 1330 ) a combination of amplitude adjustment and signal delay to create the stereo effect. For example, if the participant whose image  1205  is displayed on the left side of the displaying device  1200  is currently speaking, a delay is introduced (at  1325 ) on the right loudspeaker and the amplitude of the right loudspeaker is optionally reduced to make the signal from the left loudspeaker appear to be stronger. 
     Similarly, if the participant whose image  1215  is displayed on the right side of the displaying device  1200  is currently speaking, a delay is introduced (at  1330 ) on the left loudspeaker and the amplitude of the left loudspeaker is optionally reduced to make the signal from the right loudspeaker appear to be stronger. In contrast, if the participant whose image  1210  is displayed on the center of the displaying device  1200  is currently speaking, no adjustments are done to the signals sent to the loudspeakers. 
     Audio panning helps identify the location of the currently speaking participants on the screen and produces stereo accounting for location. In some embodiments of the invention, a delay of about 1 millisecond ( 1/1000 second) is introduced and the amplitude is reduced by 5 to 10 percent during the audio panning operation. One of ordinary skill in the art, however, will realize that other combinations of amplitude adjustments and delays might be used to create a similar effect. 
     In some embodiments, certain participant actions such as joining conference, leaving conference, etc. can trigger user interface sound effects on other participants&#39; computers. These sound effects may also be panned to indicate which participant performed the associated action. 
     In the embodiments where the focus point is also a conference participant (such as the embodiment illustrated in  FIG. 1 ), the focus point module also uses the above-described methods to present the audio for the participant whose computer serves as the conference focus point. 
     While the invention has been described with reference to numerous specific details, one of ordinary skill in the art will recognize that the invention can be embodied in other specific forms without departing from the spirit of the invention. In other places, various changes may be made, and equivalents may be substituted for elements described without departing from the true scope of the present invention. Thus, one of ordinary skill in the art would understand that the invention is not limited by the foregoing illustrative details, but rather is to be defined by the appended claims.

Metadata:
Filing Date: 20101129
Publication Date: 20130604
Grant Date: 20130604
Priority Date: 20050428
Inventors: JEONG HYEONKUK
SALSBURY RYAN
Assignee: APPLE INC
CPC Classifications: [{"code": "H04N7/15", "inventive": true, "first": true, "tree": "[]"}, {"code": "H04N7/15", "inventive": true, "first": true, "tree": "[]"}]
Family ID: 37215530