According to one embodiment, a signal processing apparatus correlates a plurality of communication terminals as a group and enables one-to-many communications in the group. The signal processing apparatus includes processing circuitry. The processing circuitry assigns a transmission right to one of the communication terminals in the group. The processing circuitry generates text data based on voice data from said one of the communication terminals in possession of the transmission right. The processing circuitry gives a texting completion notice indicative of completion of texting processing to the communication terminals in the group. The processing circuitry transmits, after the texting completion notice is given, the generated text data to at least one of the communication terminals in the group.

A method and apparatus for sound enhancement are provided in this invention. The method comprises: obtaining sound signals and converting the sound signals into digital signals; decomposing the digital signals to obtain a plurality of IMFs or pseudo-IMFs; selectively amplifying the amplitudes of the IMFs and pseudo-IMFs; reconstituting the selectively amplified IMFs or pseudo-IMFs to obtain reconstituted signals and converting the reconstituted signals into analog signals. The present invention is based on the Hilbert-Huang transform. Through the present invention, the sound can be selectively amplified, and only the high-frequency consonants in the sound are amplified without vowel, which effectively improves the clarity of the enhanced sound. The present invention overcomes the problems in the current sound enhancement method which makes the sound louder without increasing the clarity.

A disconnection of a client device is disconnected during a multi-participant communication, such as a call or a conference. An indication of the disconnection is transmitted to the client device to cause an agent at the client device to record media locally at the client device. The media recorded by the agent at the client device based on the indication of the disconnection is later received and included within a recording of the communication. For example, a gap of the recording in which the disconnection occurred may be identified, such as by performing a comparison of media within the recording to identify a start time of the gap and an end time of the gap. The media is then inserted within a portion of the recording of the multi-participant communication corresponding to the gap.

A distributed audio processing system is disclosed, for providing users with the capability of producing a personalized audio mix of a plurality of signals from a plurality of audio sources. The system includes a wireless transmitter for each audio source and, for each user, a wireless receiver. The receiver comprises a programmable audio signal processor configured to process and mix a plurality of audio tracks received via a radio broadcast of a multi-track audio signal comprising the audio signals from the plurality of sources, said processing and mixing being programmable via received commands, instructions and/or parameters. The transmitters are configured to process the audio signals received from their respective sources, according to received commands, instructions and/or parameters. According to an embodiment, a user may provide commands, instructions and/or parameters to any of the receivers and/or transmitters of the system.

Disclosed herein are systems, methods, and non-transitory computer-readable storage media for collecting call data, feeding call data to applications, and providing advanced call features.

This disclosure describes, in part, techniques for establishing network-based data communications (e.g., voice calls, video calls, etc.) between a user device of a user and a remote device of another user, and transitioning the data communications to a different user device of the user based on various types of information. In some examples, the user devices may be located in one or more environments of the user, and the data communications may be transitioned between the user devices based, at least in part, on a location of the user in the environment(s) relative to the multiple devices. For instance, if a user device is performing data communications with the remote device, but it is determined that the user has moved into a closer proximity to another user device, the performance of the data communications may be transitioned to the other user device to which the user is in closer proximity.

A system controls the audio focus for a user in a virtual conference call. The system retrieves sound parameters associated with participants in the virtual conference call with the user from a user profile of the user. The sound parameters define volume adjustments to be applied to audio data received from the participants for generating an audio mix customized to the user. The system receives the audio data from client devices of the participants, and for each of the participants, adjusts the audio data of the participant using the associated sound parameter of the participant. The system adds the adjusted audio data of the participants to the audio mix for the user and provides the audio mix to a client device of the user.

In a teleconferencing method, a first media stream and a second media stream of a teleconference are received, by processing circuitry of a first device, from a second device. The first media stream includes first audio and the second media stream includes second audio. Default weight information is received from the second device. The default weight information indicates a first audio weight for weighting the first audio and a second audio weight for weighting the second audio. The first audio weight for weighting the first audio and the second audio weight for weighting the second audio are determined based on the default weight information. Mixed audio is generated, by the processing circuitry of the first device, by combining a weighted first audio based on the first audio weight applied to the first audio and a weighted second audio based on the second audio weight applied to the second audio.

Embodiments described herein provide for the dynamic adjustment of traffic associated with audiovisual conferences or other types of communication sessions in situations where a network connection of one or more conference participants exhibits issues that may affect audio and/or video quality. The adjustment may include the determination at a particular conference participant (e.g., a particular User Equipment (“UE”)) of degraded network conditions, the generation of condensed conference information at the UE, and the outputting of the condensed conference information via the network. The condensed conference information may be included in control signaling. The condensed conference information may be used to generate reconstructed conference information, which may be distributed to other conference participants.

During an electronic call between two individuals, a sound localization point simulates a location in empty space from where an origin of a voice of one individual occurs for the other individual.