Abstract: An audio device may be configured to have multiple modes of operation. The mode of the device may be associated with a particular geometric configuration of the device, such as by swiveling an arm of the device and/or changing the orientation of the device with respect to gravity and/or a surface (e.g., table) on which the device rests. The device may determine the appropriate state based on the current geometry, and based on the state may tune the device for a particular operation suited for that geometry. For example, different signal processing parameters may be applied, different microphones may be enabled and disabled, and/or different acoustic conditions may be provided (for example, different modes of a passive radiator) based on the mode.

Abstract: A method of audio signal processing, the method comprising: accessing user-specific audio processing information for a particular user; determining identity information of an audio device for producing sound output from an audio signal; based on the identity information of the audio device, accessing device-specific audio processing information for the audio device; generating a customized audio-processing procedure for the audio signal based on the user-specific audio processing information and the device-specific audio processing information; and generating a customized audio signal by processing the audio signal with the customized audio-processing procedure.

Abstract: A method of audio signal processing, the method comprising: accessing user-specific audio processing information for a particular user; determining identity information of an audio device for producing sound output from an audio signal; based on the identity information of the audio device, accessing device-specific audio processing information for the audio device; generating a customized audio-processing procedure for the audio signal based on the user-specific audio processing information and the device-specific audio processing information; and generating a customized audio signal by processing the audio signal with the customized audio-processing procedure.

Abstract: A first access point is included in a first VLAN but not included in a second VLAN. The first access point is operatively coupled to a second access point that is included in the second VLAN but not included in the first VLAN. The second VLAN includes a multicast domain name system (mDNS) service that is not multicast to the first VLAN. The first access point is configured to receive an mDNS request for the mDNS service from a client device that is operatively coupled to the first VLAN. The first access point is configured to send, to the second access point, an encapsulated mDNS request that is based on the mDNS request from the client device such that a connection is established between the client device and a network device providing the mDNS service within the second VLAN.

Abstract: Techniques are described by which an IP telephone system leverages the digital signal processing functions of end-user IP telephones by distributing signal processing tasks typically carried out by a centralized IP-PBX. The end-user IP telephones publicize their signal processing capabilities and availabilities to an IP-PBX, which maintains a resource capability mapping of the IP telephones. When the IP-PBX receive a bitstream for a communication session involving IP telephones and/or legacy phones of the IP telephone system, the IP-PBX determines the signal processing requirements for the bitstream, selects an available, capable IP telephone to perform the requirements, and distributes the bitstream to the selected IP telephone. The IP telephone performs the requisite signal processing and returns the processed bitstream to the IP-PBX, which forwards the processed bitstream to the destination endpoint for the communication session.

Abstract: Techniques are described by which an IP telephone system leverages the digital signal processing functions of end-user IP telephones by distributing signal processing tasks typically carried out by a centralized IP-PBX. The end-user IP telephones publicize their signal processing capabilities and availabilities to an IP-PBX, which maintains a resource capability mapping of the IP telephones. When the IP-PBX receive a bitstream for a communication session involving IP telephones and/or legacy phones of the IP telephone system, the IP-PBX determines the signal processing requirements for the bitstream, selects an available, capable IP telephone to perform the requirements, and distributes the bitstream to the selected IP telephone. The IP telephone performs the requisite signal processing and returns the processed bitstream to the IP-PBX, which forwards the processed bitstream to the destination endpoint for the communication session.

Abstract: According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.

Abstract: A method and system for logging voice quality issues for a communication connection includes receiving a signal for logging quality information for a voice connection at an endpoint of the voice connection. Voice samples are collected from the voice connection at the endpoint. The voice samples are stored in an error log at the endpoint.

Abstract: A method and system for participant control of privacy during a multiparty communication session includes receiving a request from a first participant to a multiparty communication connection for a sidebar between the first participant and a second participant to the multiparty communication connection. The sidebar is provided by at least substantially eliminating voice streams generated by the first participant and the second participant from conference output streams generated for a set of remaining participants to the multiparty communication connection.

Abstract: A network device and its mode of operation. The network device includes an input port through which audio streams may be received. A controller receives a user input indicating which type of mix the user wants applied to the audio stream and applies the mixing factor designated. The mixing factor may be a stereo mix, a panned mix, or a mono mix. The network device also includes an output port through which the mixed audio stream can be transmitted to the user.

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 method is disclosed, where the RX unit of an IP telephone waits for a first silence indication packet to be received. When the first silence indication packet is received, the RX begins attenuating incoming packets. The rate of attenuation may occur at a rate responsive to an indication provided by the TX unit. The RX unit also mixes comfort noise with the attenuated incoming packets. The RX unit the waits for a second silence indication packet to be received. When the second silence indication is received the RX unit mixes in comfort noise at a level responsive to an indication provided by the TX unit. The RX units then waits for voice packets to be received from the TX unit. When normal voice activity resumes, the RX unit stops the generation of comfort noise and begins playing voice packets again.

Abstract: A method and apparatus for improving the mute function of an IP telephone is disclosed, where the method comprises determining, by a transmitting IP telephone, whether the mute function has been activated; and if the mute function has been activated, then providing, a predetermined output. Various aspects of the method are disclosed, including: an aspect where the predetermined output is characterized by the absence of silence as detected by an RX unit when the mute function of a TX unit is activated; an aspect where the predetermined output comprises a signal in which voice information has been removed; and an aspect where the removing of voice information is performed by an audio filter disposed within the TX unit of an IP telephone.