摘要:
The present invention provides methods, devices, and systems for creating an ad-hoc conference station from speakers on a plurality of mobile communication devices. The ad-hoc conference station can cancel the echo of any incoming audio for all of the communication devices. This feature may be achieved by employing a master/slave configuration for the communication devices within a common area.
摘要:
Tonal correction of speech is provided. Received speech is analyzed and compared to a table of commonly mispronounced phrases. These phrases are mapped to the phrase likely intended by the speaker. The phrase determines to be the phrase the user likely intended can be suggested to the user. If the user approves of the suggestion, tonal correction can be applied to the speech before that speech is delivered to a recipient.
摘要:
The automatic completion of composite characters is supported by the generation of lists of candidate words or characters. Such lists may be generated by specifying letters or word shapes that are required to be included in candidate words or characters, independent of the order in which a specified letter or word shape is traditionally added to the completed word or character. In a subtractive mode, a user may exclude words or characters that include one or more letters or word shapes specified by the user.
摘要:
The present invention provides a communication monitoring and analysis method and system. More specifically, the present invention provides a method for determining the health and overall satisfaction of employees in an organization. The determination may be made by monitoring communications generated by employees for their tone and other parameters related to their satisfaction with various decisions made within the organization.
摘要:
The present invention provides a communication monitoring and analysis method and system. More specifically, the present invention provides a method for determining the health and overall satisfaction of employees in an organization. The determination may be made by monitoring communications generated by employees for their tone and other parameters related to their satisfaction with various decisions made within the organization.
摘要:
The present invention provides a communication monitoring and analysis method and system. More specifically, the present invention provides a method for determining the health and overall satisfaction of employees in an organization. The determination may be made by monitoring communications generated by employees for their tone and other parameters related to their satisfaction with various decisions made within the organization.
摘要:
A method and apparatus for testing network performance are provided. In data provided by an application on a first host for transport or communication to an application associated with a second host according to a first data transport protocol is intercepted at the first host and wrapped or encapsulated in a test data packet formatted according to a second data transport protocol. The test data packet formatted according to the second data transport protocol includes, in addition to data comprising all or a portion of the original data packet, instrumentation information. The test data packet is then delivered to the second host, which unpacks the original data packet and the instrumentation information. A response packet containing instrumentation information may be sent from the second host to the first host to provide roundtrip performance metrics.
摘要:
The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.
摘要:
The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.
摘要:
The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.