Abstract: A host device of a serial device system having a serial communications connection The host device including a host serial communication transceiver connected to a peripheral serial communication transceiver of a peripheral device via a serial connection of a serial communications cable having communication lines. The host device measures electrical power consumption of the host serial communication transceiver to determine when serial communications is prevented due to the serial communications cable being disconnected which includes at least one of the communication lines being detached from the host, detached from the peripheral device, or broken.

Abstract: An ordering system can be positioned partially, or completely, outside in a retail environment with an ordering device located outside of a building on a site. The ordering device receiving a first audio stream concurrently with a second audio stream from an employee and proceeds to capture the first audio stream with a first port of an on-site computing device while capturing the second audio stream with a second port of the on-site computing device. A customer strategy can be executed with an intelligence module of the on-site computing device connected to the ordering device with the on-site customer strategy directing automated interactions with a first on-site customer to compile a retail order. The employee may communicate directly with the intelligence module via the second port without interrupting the first audio stream.

Abstract: A visitor detection, facial recognition, and alert system and processes for assisting memory-challenged patients to recognize entryway visitors are disclosed. The visitor detection, facial recognition, and alert system and processes assist a memory-challenged patient to distinguish between known and unknown visitors and inform a caretaker of such known and unknown visitors. The visitor detection, facial recognition, and alert system provides an assistive tool for Alzheimer's patients to recognize their visitors using facial recognition technology.

Abstract: Example embodiments may include one or more of receiving sound emissions signals from channels via sound emitters, controlling the sound emission signals, via relay circuits, and one of the relay circuits is configured to interrupt one of the sound emission signals associated with one of the sound emitters while the other sound emissions signals pass to the other corresponding sound emitters, and receiving, via a sound detection circuit, an electrical ambient sound signal based on ambient sound sensed by the one of the sound emitters responsive to the interrupted one of the sound emission signals.

Abstract: A double-talk state can be accurately detected, and based on a detection result, echo can be appropriately suppressed. When a sound is output from a speaker and only the output sound is input to a microphone, a comparison is made, for each of different frequency bands, between a frequency mask generated based on a power spectrum or an amplitude spectrum for a learning signal transmitted through a transmitting signal path and a value of a power spectrum or an amplitude spectrum for an input signal input from the microphone, to detect whether there is a double-talk state. In a case of detecting that no signal is being transmitted through the transmitting signal path and that a signal is being transmitted through the receiving signal path, an echo suppressor is used to execute processing of suppressing an echo in the input signal.

Abstract: A system includes participant nodes that unicast a voice stream to a master node and receive a multicast voice stream from the master node. The multicast voice stream may be a mixed voice stream mixing the voice streams of the participants. In this case, each participant performs echo cancellation on the multicast voice stream. Alternatively, the multicast voice stream may be a multiplexed voice stream multiplexing the voice streams of all participants. In this case, each participant de-multiplexes the multicast voice stream and generates a mixed voice stream that does not include that participant. Alternatively, the multicast voice stream received by each participant may be a multiplexed digital voice stream multiplexing multiple mixed streams including one mixed stream that does not include the voice stream of that participant. In this case, each participant de-multiplexes the multicast voice stream to obtain the mixed stream that does not include that participant.

Abstract: Automotive acceleration compensation force sensing includes determining a total force value from the measurements from force sensors, obtaining, from an accelerometer, an environmental acceleration value, and calculating an input object force value on an input surface using a total force value and the environmental acceleration value. Automotive acceleration compensation force sensing further includes reporting the input object force value.

Abstract: Systems and methods for automatically adjusting telephonic conversation volume are provided. A computer-implemented method includes: detecting a telephone being used for a conversation and receiving data from at least one of a microphone, a camera, and a location system associated with the telephone. The method includes analyzing the data to determine at least one of: usage of sensitive or confidential words in the conversation; distance of the telephone to another person; level of annoyance of another person; conversation loudness relative to ambient loudness; location of the telephone relative to quiet areas. The method includes automatically adjusting a volume of a speaker of the telephone during the conversation based on the analyzing.

Abstract: An entertainment audio processing system includes a music limiter configured to receive an audio signal, volume control information for music, and volume control information for announcement. The music limiter is further configured to calculate a gain parameter based on a music-to-announcement ratio. In another aspect, an entertainment audio processing system includes a mixer configured to substantially match a left audio signal channel and a right audio signal channel. The entertainment audio processing system further includes an equalizer configured to introduce magnitude and phase changes between the left audio signal channel and the right audio signal channel.

Abstract: Systems and methods for automatically adjusting telephonic conversation volume are provided. A computer-implemented method includes: detecting a telephone being used for a conversation and receiving data from at least one of a microphone, a camera, and a location system associated with the telephone. The method includes analyzing the data to determine at least one of: usage of sensitive or confidential words in the conversation; distance of the telephone to another person; level of annoyance of another person; conversation loudness relative to ambient loudness; location of the telephone relative to quiet areas. The method includes automatically adjusting a volume of a speaker of the telephone during the conversation based on the analyzing.

Abstract: Various embodiments are directed to cooperation among communications devices having microphones to employ their microphones in unison to provide voice detection with noise reduction for voice communications. A first communications device comprises a processor circuit; a first microphone; an interface operative to communicatively couple the processor circuit to a network; and a storage communicatively coupled to the processor circuit and arranged to store a sequence of instructions operative on the processor circuit to store a first detected data that represents sounds detected by the first microphone; receive a second detected data via the network that represents sounds detected by a second microphone of a second communications device; subtractively sum the first and second data to create a processed data; and transmit the processed data to a third communications device. Other embodiments are described and claimed herein.

Abstract: An audio signal processing device includes a volume setting unit that changes a volume of an audio signal according to a control for changing the volume, and a level control unit that controls a level of a predetermined band of the audio signal according to the control.

Abstract: Systems and methods for automatically adjusting telephonic conversation volume are provided. A computer-implemented method includes: detecting a telephone being used for a conversation and receiving data from at least one of a microphone, a camera, and a location system associated with the telephone. The method includes analyzing the data to determine at least one of: usage of sensitive or confidential words in the conversation; distance of the telephone to another person; level of annoyance of another person; conversation loudness relative to ambient loudness; location of the telephone relative to quiet areas. The method includes automatically adjusting a volume of a speaker of the telephone during the conversation based on the analyzing.

Abstract: The present disclosure is directed to devices, methods and systems for microphone arrays wherein enhancing performance of directional microphone arrays is provided. Enhanced performance of speaker phones is also provided. In certain embodiments, the housing of the device is configured to support the at least three microphones and the loudspeaker in a substantially first orientation; and the at least three microphones and the loudspeaker are arranged in a spatial relationship such that appropriate phase and delay characteristics achieve a substantial null response in the at least three microphones and in the loudspeaker in a substantial vertical direction away from the substantially first orientation over a desired audible range of frequencies and the device is able to provide a response to sounds over a range of first oriented elevations.

Abstract: A multivoice radio system allows users to speak and listen, at the same time, to others using multivoice radios without using a base station. Several embodiments of the present invention are directed toward combining a multivoice radio with push-to-talk (PTT), long-range radio. Other embodiments combine two or more multivoice systems and/or use an extender to increase a distance that multivoice radios can communicate with each other. Combining multivoice systems, in some embodiments, creates a voice and/or data mesh network. Switch-to-talk (STT) functionality can be added to systems to increase a number of users and promote radio discipline. Additionally, a combined in-ear microphone and speaker is disclosed. Further embodiments are directed to combining duplex radios, such as mobile phones, to multivoice systems, and/or PTT systems.

Abstract: A controller for an echo suppressor configured to suppress a residual echo of a far-end signal included in a primary error signal, the controller adapted for operation with a primary adaptive filter configured to form a primary echo estimate of the far-end signal included in a microphone signal and an echo canceller configured to cancel that primary echo estimate from the microphone signal so as to form the primary error signal, the controller comprising: a secondary adaptive filter configured to form a secondary echo estimate of the far-end signal comprised in the microphone signal; and control logic operable in at least two modes selected in dependence on a convergence state of the primary adaptive filter, the control logic being configured to control activation of the echo suppressor in dependence one or more transient or steady state decision parameters.

Abstract: A hand-held electronic device includes an earpiece, a first speaker having a first primary audio output ported to the earpiece, a second speaker having a second primary audio output and a second bass reflex output, with the second bass reflex audio output also being ported to the earpiece. The device includes a use case manager configured to determine a mode of operation of the device, and a controller configured to drive the first and second speakers in either a broadcast mode or a private mode in response to a signal from the use case manager. The polarity of the signal driving the second speaker is inverted for frequencies below the bass reflex port resonance frequency, to thereby provide a higher total combined energy in the vicinity of port resonance.

Abstract: An electronic device may be coupled to an accessory such as a pair of earphones. The earphones may have noise cancellation features that may be implemented using noise cancellation circuitry in the earphones or in the electronic device. The earphones may have ear presence sensor structures that determine whether speakers in the earphones are present at the ears of a user. In one suitable embodiment, control circuitry in the earphones may be used to adjust noise cancellation circuitry in the earphones based on information from the ear presence sensor structures. For example, the control circuitry may deactivate noise cancellation circuitry in response to receiving information from the ear presence sensor structures indicating that the earphones have been removed from a user's ears. In another suitable embodiment, control circuitry in the electronic device may adjust noise cancellation circuitry in the electronic device based on information from the ear presence sensor structures.

Abstract: Apparatus, systems including an electrical touch screen sensor and methods for implementing an electrical touch screen sensor are disclosed. For example, an apparatus is disclosed including a display having an array of display pixels and a plurality of sensors distributed amongst the display pixels and embedded within the display, where the sensors act to detect changes in capacitance associated with objects placed in proximity to the display pixels. Other implementations are also disclosed.

Abstract: Method, user device and computer program product for suppressing echo. An audio signal is output from a speaker. A microphone receives an audio signal, wherein the received audio signal includes an echo resulting from said outputted audio signal. A Finite Impulse Response filter estimate ?(t) is dynamically adapted in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(t) of the echo in the received audio signal. At least one power response is determined from the filter estimate ?(t) and used to estimate the echo power of the echo in the received audio signal. The estimated echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal.

Abstract: The present disclosure is directed to devices, methods and systems for microphone arrays wherein enhancing performance of directional microphone arrays is provided. Enhanced performance of speaker phones is also provided. In certain embodiments, the housing of the device is configured to support the at least three microphones and the loudspeaker in a substantially first orientation; and the at least three microphones and the loudspeaker are arranged in a spatial relationship such that appropriate phase and delay characteristics achieve a substantial null response in the at least three microphones and in the loudspeaker in a substantial vertical direction away from the substantially first orientation over a desired audible range of frequencies and the device is able to provide a response to sounds over a range of first oriented elevations.

Abstract: A mobile terminal capable of recognizing at least one of the presence or absence of a user and a location of the user are provided. The mobile terminal includes: a terminal main body; a user tracking unit configured to recognize at least one of the presence or absence of a user and a location of the user within a certain range based on the terminal main body; and a controller configured to generate a control signal for controlling a function of the terminal based on at least one of the presence or absence of the user or the user location.

Abstract: A communication device includes an upper housing and a bottom housing connected to each other, a master Printed Circuit Board (PCB) for realizing a communication function, a speaker and a keypad assembly. The master PCB board is arranged on the bottom housing, wherein the upper housing includes a front housing and a front cover, the keypad assembly is arranged between the front housing and the front cover; the speaker is arranged under the front housing, the keypad assembly is connected to the master PCB board by a flexible circuit board which passes through the front housing; voice spreading holes connecting to the front cavity of the speaker are correspondingly arranged on the front cover, the keypad assembly and the front housing; the joints among the front housing, the keypad assembly, the front cover and the speaker are waterproof.

Abstract: One feature discloses a method operational at a multimedia mobile call transfer device, the method comprising: receiving a first video data associated with a multimedia call from a mobile device; receiving a first audio data associated with the multimedia call from the mobile device; processing the received first video data and the received first audio data; transmitting the processed first video data to a multimedia device for display on the multimedia device; and transmitting the processed first audio data to the multimedia device for play on speakers of the multimedia device.

Abstract: A method for maintaining transmit audio quality under harsh environmental conditions, the method includes receiving audio signals into a microphone of a portable communication device and determining at least one parameter associated with the received audio signals. In accordance with an embodiment, the at least one parameter is compared with a received audio parameter threshold. When the at least one parameter falls outside of the received audio parameter threshold, the audio routing is switched from the microphone to the loudspeaker. Subsequent communication may revert back to the microphone or remain at the loudspeaker depending on monitored audio conditions.

Abstract: A sensor network system including node devices connected in a network via predetermined propagation paths collects data measured at each node device to be aggregated into one base station via a time-synchronized sensor network system. The base station calculates a position of the signal source based on the angle estimation value of the signal from each node device and position information thereof, designates a node device located nearest to the signal source as a cluster head node device, and transmits information of the position of the signal source and the designated cluster head node device to each node device, to cluster each node device located within the number of hops from the cluster head node device as a node device belonging to each cluster. Each node device performs an emphasizing process on the received signal from the signal source, and transmits an emphasized signal to the base station.

Abstract: A telephone includes: a housing; a voice output device that is placed inside the housing and produces voice; a contact detection unit that detects a position of an object that contacts the housing; a position offset calculation unit that calculates a distance between the contact position of the object detected by the contact detection unit and the voice output device; and a voice adjustment unit that adjusts the voice produced from the voice output device, depending on the distance.

Abstract: Disclosed are an apparatus and a method for voice processing in a mobile communication terminal. A plurality of microphones are used to remove environmental noise at the time of voice communication, so that it is possible to perform high-quality voice communication and video telephony. Moreover, it is possible to perform voice recording even when a user does not open a mobile communication terminal. Furthermore, when voice is recorded or sound is recorded during moving image photographing, a plurality of microphones are effectively utilized to achieve good-quality recording and to perform recording conveniently even when the folder or the slider of the mobile communication terminal is closed. Therefore, it is possible to provide improved convenience in using the mobile communication terminal.

Abstract: A voice communication end device performs quality checks to determine whether acoustic echo cancellation would be ineffective, such as due to noise or clock drift or discontinuities between incoming and outgoing voice channels. In the case where echo cancellation would prove ineffective, the device falls back on a tri-state voice switching operation that includes a bi-direction state in which both channels are on in full duplex operation, which provides a smoother transition switching between active channels. The tri-state voice switching supports both voluntary transitions where the active user voluntarily stops to yield the active channel, and forced transitions where the active user is forcedly interrupted by the other user speaking more loudly.

Abstract: In various embodiments, a speakerphone may comprise multiple (e.g., 16) microphones placed in a circular array around a central speaker. The microphones may be mounted vertically in the speakerphone with their respective diaphragms substantially parallel to the top surface of the speakerphone. The centrally mounted speaker may be coupled to a stiff internal speaker enclosure. The speaker enclosure may be made of a stiff, heavy material (e.g., a dense plastic) to prevent the speaker vibrations from excessively vibrating the speakerphone enclosure (which may affect the microphones). The speaker enclosure may also include a raised rim and ridges to increase its stiffness.

Abstract: A device comprising a circuit for picking up acoustic signals with the microphone; a circuit for playing back audio signals with an amplifier and a loudspeaker; an interface circuit for interfacing with a telephone network or a cell phone; and a circuit for digitally processing audio signals and reducing the acoustic echo that results from interaction of the loudspeaker with the microphone. The circuit for digitally processing audio signals comprises: an echo cancellation stage with an adaptive linear filter suitable for subtracting from the signal picked up by the microphone a reference signal that is derived from the signal received at the output from the interface; a gain control stage for suppressing the residual echo, and a stage for selectively reducing the background noise present in the signal received at the output from the echo suppression stage.

Abstract: The device comprises a microphone for detecting a speech signal from a near speaker, and a loudspeaker for reproducing a speech signal from a remote speaker. The processing for canceling the interfering acoustic echo implements an adaptive linear filtering algorithm. Double talk situations are detected by: evaluating an index representative of the convergence or divergence of the algorithm; assessing a predetermined condition for detecting a double talk situation; and if the condition is satisfied, modifying at least one parameter of the algorithm in response to the detection. The representative index may be the norm of the gradient vector describing the adaptation of the filter from one iteration of the algorithm to the next, the conditions being a comparison between the gradient and a threshold. The parameter that is modified double talk situation may be the adaptation stepsize of the algorithm, and also the gain control of an echo suppression stage.

Abstract: A method and system for managing a communication session is provided. The communication session is associated with multiple communication devices. The method includes learning (304) a set of derived acoustic features of an audio communication signal that is associated substantially only with one user of a communication device. The method also includes receiving (306) a communication session signal. The communication session signal is an audio signal that includes a combination of audio communication signals. Each audio communication signal of the audio communication signals is associated with a user of a communication device of the multiple communication devices. The method includes modifying (308) the communication session signal based on the set of derived acoustic features.

Abstract: A speakerphone having improved echo cancellation and sound output includes at least one directional microphone having at least one axis of sensitivity and at least one zone of insensitivity, and a speaker disposed in the zone of insensitivity of the microphone to radiate sound away from the microphone and towards a reflective surface, such as a desktop or wall, against which the speakerphone is disposed. A baseplate disposed adjacent to the speaker outlet can combine with the housing of the phone to form a flaring, right-angled horn having an inlet coupled to the outlet of the speaker and an outlet terminating at a periphery of the housing. A wall-mounting embodiment incorporates a unidirectional microphone with an axis of sensitivity oriented perpendicular to the wall, and a desktop-mounting embodiment includes an array of at least two bi-directional microphones having respective axes of sensitivity oriented parallel to the desktop.

Abstract: A method for data transmission between participants communicating by means of a data bus, in which as high a transmission likelihood for additional data as possible is to be assured by providing that in the exchange of data telegrams between the participants within the cycle time 13a predetermined for one transmission cycle 13, at least one first and at least one second time slot 11, 12 are used, and the first time slot 11 serves to forward process data and the second time slot 12 serves to forward additional data, and the length of the data telegrams representing the additional data is predeterminable within the length of the corresponding time slot 12.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: A radio communication system having a plurality of mobile stations, at least one base station, and a control side apparatus connected to the base station. Each mobile station has a mode switching instruction reception unit for receiving an instruction for switching from a radio communication mode for performing a radio communication operation to a loudspeaker mode or a hot mic mode, a destination identifying information storage unit for storing information for identifying a communication apparatus which serves as a communication partner when the mode switching instruction reception unit receives the instruction for switching to the hot mic mode, and a destination identifying information transmission unit responsive to the received instruction for switching to the hot mic mode by the mode switching instruction reception unit for transmitting the information stored in the destination identifying information storage unit over the air.

Abstract: A portable electronic device that automatically detects its mode of use and alters a characteristic of its user interface to match the mode of use. A mobile telephone is described as an example of a portable electronic device. The telephone is equipped with a proximity sensor that may indicate whether the telephone is held close or far from the user's face. Based on the proximity to the user's face, gains in the audio interface are adjusted. The gain is reduced when the device is held close to a user's face and increased when the device is moved away from the user's face. This approach is well suited for use in connection with portable devices that use voice command interfaces.

Abstract: A method is disclosed that enables the managing of the overall sound level in an enterprise environment where telephones are used. A data-processing system such as a private branch exchange monitors whether one or more telephones are in use. Based on detecting when a first endpoint is in use and, therefore, when the endpoint's user is present, the private branch exchange controls one or more characteristics of the loudspeaker volume at a second endpoint. By accounting for other considerations such as the spatial closeness between the endpoints, which can be determined from office dimensions stored in a database, the private branch exchange of the illustrative embodiment is able to determine the degree of sound that is coupling over from one endpoint location to another. On a larger scale, the exchange is able to control the loudspeaker volumes of all of the endpoints in the workplace area. In doing so, the exchange manages the overall acoustic noise present.

Abstract: The present invention discloses a volume control process of a mobile device to resolve issue of using the same volume between the speech state and the ring state in a traditional mobile device. Speaker volume output of the mobile device in the ring state and the speech state is set by a ring volume mode and a speech volume mode, respectively.

Abstract: The present invention is directed to a table that is constructed to receive an audio system and an audio system that is intended to be incorporated within a tabletop. The audio system comprises a housing and an audio column having a speaker wherein the audio column is capable of being retracted into the housing such that the top wall of the audio column is substantially co-planar with a tabletop. Additionally, the present invention includes a method of conducting business meetings including releasing a mechanism causing the audio column to extend out of the housing; allowing sound to travel through the speaker during the meeting and to be heard by participants in a room where the tabletop is located; and closing the audio column such that the audio column is received within the housing and the top of the audio column is at least substantially co-planar with the tabletop.

Abstract: A telephone having a PA system includes a telephone set, a handset, a conversion module, and an external loudspeaker. The telephone set is electrically connected to the handset via the conversion module. The external loudspeaker is electrically coupled to the conversion module.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: An electronic device having a first piece, a second piece moveable with respect to the first piece, and a position sensor operable to sense the position of the second piece relative to the first piece. The electronic device is operated in a first mode or a second mode dependent upon the position of the second piece relative to the first piece. Optionally, a proximity sensor is provided to detect the presence of a body in close proximity to the device when the second piece is in an open position relative to the first piece.

Abstract: Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network. The decoding device uses the relative delay to drive a playout splitter—once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference.

Abstract: A system and a method for the transmission and reception of digital data wirelessly using acoustic tones comprises an encoder, a data interleaving system, an error correction system, and a decoder. The digital data is decoded by using in-phase and quadrature filters and through a score based function that determines the data that was most likely transmitted.

Abstract: Communication terminals, methods, and computer program products are provided that sense background noise via a speaker that is also used to generate sounds. In some methods of operating a communication terminal, and a speaker signal is supplied to a first speaker to generate sound therefrom, a noise sensing signal is received from the first speaker. The noise sensing signal includes a contribution associated with background noise that is incident to the first speaker. Presence of the background noise in the noise sensing signal is determined.

Abstract: A signal conditioning circuit is described which can facilitate dual mode use of a connector circuit, e.g., audio mode or data mode. In other words, a headset or alternatively, a data interface may be coupled to the connector circuit and signal conditioning circuit, depending on the mode of the device. In accordance with this disclosure, the signal conditioning circuit operates in audio mode without the need for conventional blocking capacitors in the speaker channels. Moreover, in some embodiments, the signal conditioning circuit also provides backwards compatibility with conventional connector circuits that include such blocking capacitors.

Abstract: A pocket speakerphone adapter for a landline-based common telephone to provide the common telephone with speakerphone capabilities. The conventional handset of a conventional common telephone is disconnected from the handset jack of its base unit, and the pocket speakerphone is temporarily connected in its place to the handset jack of the base unit using a conventional handset cord. The conventional common telephone base unit provides interconnection with the public switched telephone network (PSTN), while the pocket speakerphone in accordance with the principles of the present invention provides a temporary adaptation of the land line common telephone to operate as a speakerphone.

Abstract: Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network. The decoding device uses the relative delay to drive a playout splitter—once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference.