Abstract: Disclosed is a headphone earcup having a frame with an inside surface to be positioned adjacent a user's head, an outside surface opposite the inside surface, and a perimeter surface. A central hole in the frame extends from the inside surface to the outside surface. A removable cover on the outside surface of the frame covers the central hole in the frame. A first wave guide holds an earbud. The first wave guide has a perimeter surface and is sized in relative proportions to nest in the central hole of the frame. A second wave guide is positioned on the inside surface of the frame and is covered by a ported cover. A padded cover is positioned over the cover of the second wave guide for comfort.

Abstract: A video is received from a video capture device. The video capture device has a front facing camera and a display screen which displays the video captured by the video capture device in real time to a user. One or more images of a pinna and head of the user in the video are used to automatically determine one or more features associated with the user. The one or more features include an anatomy of the user, a demographic of the user, a latent feature of the user, and an indication of an accessory worn by the user. Based on the one or more features and one or more HRTF models, a head related transfer function (HRTF) is determined which is personalized to the user.

Abstract: Systems and methods are provided for generating application data from call data. In one implementation, a method includes acquiring call data from a call-data source with a call-data aggregator; modifying a portion of the call data with a call-data modifier; and generating application data from the portion of the call data. Application data may be configured for diagram generation. The diagram may graphically indicate call volume in branches of an interactive voice response (IVR) system map. The diagram may be a flow diagram including a connector associated with a branch of the IVR system map and the connector may have a width proportional to a call volume in the branch of the IVR system map. The diagram may indicate call-portion durations, which may be associated with a phase of a call and which may have a color associated with a phase of a call.

Abstract: In aspects, systems, methods, apparatuses and computer-readable storage media implementing embodiments for mixing audio content based on a plurality of user generated recordings (UGRs) are disclosed. In embodiments, the mixing comprises: receiving a plurality of UGRs, each UGR of the plurality of UGRs comprising at least audio content; determining a correlation between samples of audio content associated with at least two UGRs of the plurality of UGRs; generating one or more clusters comprising samples of the audio content identified as having a relationship based on the determined correlations; synchronizing, for each of the one or more clusters, the samples of the audio content to produce synchronized audio content for each of the one or more clusters, normalizing, for each of the one or more clusters, the synchronized audio content to produce normalized audio content; and mixing, for each of the one or more clusters, the normalized audio content.

Abstract: A method for communicating with a call center, is performed at a device having one or more processors and memory. The device establishes, by an audio and video client, a network session connection to the call center by using an audio and video server. The device receives, through the network session connection, a first-level voice menu sent by the call center. The device obtains a correspondence between visualized menus and voice menus of the call center. The device searches for a first visualized menu corresponding to the first-level voice menu according to the correspondence. The device presents the first visualized menu in a user interface of the audio and video client.

Abstract: A real-time messaging system platform receives from, a communication application installed on a first user communication device, a first user identifier, an indication that a messaging service message composed by the first user is being directed to a first destination associated with a second user, and an identification of data present in the messaging service message. The identification of the data and a message transmission history of the first user is used to enable selection of a message of a first entity. The selected message of the first entity is caused to be displayed by the communication application within a message framework that frames the first user messaging service message. The first entity message is caused to be included in the messaging service message transmitted to the second user device, wherein the first entity is not provided with access to the identity of the first user or the second user.

Abstract: A method for manufacturing an electrodynamic transducer from a membrane system module and a magnet system module is disclosed. The membrane system module comprises an annular first chassis unit made by injection-molding and a diaphragm fixed thereon. A coil is fixed on a coil seat of the diaphragm. The magnet system module comprises a pole piece, a magnet and a yoke that each have a central hole, and a second chassis unit that is made by injection-molding and that fills the central holes in the pole piece, the magnet and the yoke. It surrounds the yoke at least partially, so that the pole piece, the magnet and the yoke are held. In the transducer, the membrane system module and the magnet system module are plugged together, wherein an annular recess on the lower side of the first chassis unit is arranged on a circumferential shoulder of the second chassis unit. This defines a position of the coil relative to the pole piece, to the magnet and to the yoke of the magnet system module.

Abstract: Techniques are provided for reduction of noise and nonlinear-echo. A methodology implementing the techniques according to an embodiment includes estimating transfer functions (TFs) of echo paths of audio signals received through a microphone array. The audio signals include speech signal, additive noise, and echo, the TF estimation based on the reference signal. The method also includes cancellation of linear components of the echo, based on the echo path TFs. The method further includes estimating an inverse square root of a covariance matrix of the additive noise, whitening the echo cancelled signals, and estimating a speech path RTF associated with the speech signal, based on the whitened echo cancelled signals. The method further includes performing beamforming on the whitened signals (such as weighted MVDR beamforming), based on the echo path TFs, the speech path RTF, and the estimated inverse square root additive noise covariance matrix.

Abstract: A switch device for use in a gateway equipment and a gateway equipment comprising the switch device are suggested. The suggested switch comprises a Foreign Exchange Station (FXS) port configured to be connected to a telephone set, a Foreign Exchange Office (FXO) port connected to communicate analog voice signals with an external network, and a Voice over Internet Protocol (VOIP) unit configured to communicate digital data signals with the external network and enable a VOIP call using the digital data signals. The switch device comprises a switch configured to, when the gateway equipment is power down, connect the FXS port to the FXO port. The switch device also comprises a power unit configured to generate a work voltage from the analog voice signals at the FXO port to provide a power supply to the switch.

Abstract: This document discusses, among other things, limiting contact to a networked contact center that is a host to multiple tenants. In an example embodiment, a communication is received by a networked contact center. In determining whether to allow the communication to reach contact center resources, a contact rate value may be examined. The contact rate value may represent an allowable quantity of contact within a time interval. For some example embodiments, a determination of whether the network contact center is to accept or reject the communication is made based on the contact rate value.

Abstract: Improved translation operations are disclosed. A first segment of text in a first language is received, to be translated into a second language. The first segment is evaluated, by operation of one or more processors, of text using the pattern model to generate a first s-score. A second segment of text in the second language is generated based on processing the first segment of text using a machine translation model. Upon determining that the first s-score exceeds a predefined threshold, the first segment of text is flagged for auto-substitution with the second segment of text, such that the first segment of text is not reviewed by a human editor.

Abstract: An audio device, which is configured to receive audio data from anther device, comprises a memory that stores instructions and a processor. The processor is configured to execute the instructions stored in the memory to: receive the audio data and a time stamp attached to the audio data, output a pulse data having a predetermined length, detect end time of the pulse data, compare the end time of the pulse data and the time stamp, adjust the audio data based on the comparing result, and output the adjusted audio data.

Abstract: A system, method, and computer program product are provided for translating unified ticketing system (UTS) messages. In operation, a translation system receives one or more messages in at least one first language being communicated between a customer ticketing system and a unified ticketing system associated with a communication service provider. The translation system translates the one or more messages, including any corresponding attachments and screenshots, from the first language to at least one second language. Further, the translation system releases the translated one or more messages, including the corresponding translated attachments and screenshots, to be communicated between the customer ticketing system and the unified ticketing system.

Abstract: Disclosed herein a method of and system for providing a summary of a document. The method may comprise identifying, using a processor, a plurality of salient excerpts from the document. Further, the method may comprise generating, using the processor, the summary of the document based on the plurality of salient excerpts. The method may comprise receiving at least one parameter configured to control the generating of the summary. Accordingly, identifying or generating may be based on the at least one parameter. The method may further comprise identifying a plurality of key terms in the document. Moreover, the method may include ranking each sentence of the document based on the plurality of key terms. Additionally, the method may comprise identifying a set of key sentences based on the ranking. Further, the set of key sentences may constitute the plurality of salient excerpts.

Abstract: Disclosed herein is method and system for identifying one or more Places of Interest (PoI) in a natural language system. Word embedding representation for each word in the natural language input are retrieved from a knowledge repository. Further, for each word, Part-of-Speech (POS) tags are tagged, and dependency labels are generated. Subsequently, a PoI tag is assigned to each word based on the word embedding representation, the POS and the dependency labels of each word. Finally, the one or more PoI are identified based on PoI tag assigned to each word. In an embodiment, the method of present disclosure helps in dynamically identifying one or more PoI from natural language text utterances in interactive systems, thereby enhancing usability of interaction based intelligent systems.

Abstract: [Object] To provide an information processing system and an information processing method capable of auditing the utterance data of an agent more flexibly. [Solution] An information processing system including: a storage section that stores utterance data of an agent; a communication section that receives request information transmitted from a client terminal and requesting utterance data of a specific agent from a user; and a control section that, when the request information is received through the communication section, replies to the client terminal with corresponding utterance data, and in accordance with feedback from the user with respect to the utterance data, updates an utterance probability level expressing a probability that the specific agent will utter utterance content indicated by the utterance data, and records the updated utterance probability level in association with the specific agent and the utterance content in the storage section.

Abstract: An automatic translation device includes a communications module transmitting and receiving data to and from an ear-set device including a speaker, a first microphone, and a second microphone, a memory storing a program generating a result of translation using a dual-channel audio signal, and a processor executing the program stored in the memory. When the program is executed, the processor compares a first audio signal including a voice signal of a user, received using the first microphone, with a second audio signal including a noise signal and the voice signal of the user, received using the second microphone, and entirely or selectively extracting the voice signal of the user from the first and second audio signals, based on a result of the comparison, to perform automatic translation.

Abstract: A system and method of a telephony application platform can include receiving a communication session request specifying a destination endpoint, connecting the communication session through a first application module according to the destination endpoint, receiving an instruction to transition communication control to a second application module, transitioning communication control of the communication session to the second application module, independently metering resource usage of the communication session of each application module, and accounting the metered usage of the communication session.

Abstract: Systems and methods for the matching of datasets, such as input audio segments, with known datasets in a database are disclosed. In an illustrative embodiment, the use of the presently disclosed systems and methods is described in conjunction with recognizing known network message recordings encountered during an outbound telephone call. The methodologies include creation of a ternary fingerprint bitmap to make the comparison process more efficient. Also disclosed are automated methodologies for creating the database of known datasets from a larger collection of datasets.

Abstract: A panel vibration type sound generating display device is disclosed. The display device includes a display panel for displaying an image; a sound generating actuator connected to the display panel and configured to vibrate the display panel to generate sound; a support structure spaced apart from the display panel with an air gap between the support structure and the display panel; and a baffle part disposed between the support structure and the display panel to surround the air gap.