Paltalk Holdings, Inc.

Other

8.times.8, Inc.

Netergy Networks, Inc.

Netergy Microelectronics, Inc.

Katherine Wang Kwan

Paul A Voois

Frank Chu

Barry Dean Andrews

Ruili Hu

May Shu-Pei Chiang

Stephane Bryant

Bryan R Martin

Paul Ning

Paul C Ning

Daniel W Wright

Corey Gates

Virgil Patrick Dobjanschi

A method and computer program product which allows both phone-based and IP-based clients to participate in a single audio conference. The method enables at least two multi-point control units (MCUs) (i.e., conferencing servers) to connect via a standard data linkage (i.e., full-duplex dial-up or IP link). The method and computer program product enables the phone-based MCU to handle the phone clients and the IP-based MCU to handle the IP-based clients, while connecting the two to allow each participating client to hear all other participating clients.

System architecture for linking packet-switched and circuit-switched clients

A method and computer program product allows for the efficient allocation of buffers (e.g., first-in first-out (FIFO) queues) for current and predicted active speakers in voice conferencing systems. The method and computer program product, implemented by a server hosting an audio conference for a plurality of speakers, minimizes the loss of audio data for speakers as they switch from 'non-active' to 'active' status. This is accomplished by employing a set of active speaker buffers and a set of predicted active speaker buffers. The predicted active speaker buffers maintain a collection of the most recent x packets or m milliseconds of &#x201c;non-active&#x201d; speaker audio data, and transfer a portion of the data from the predicted active speaker buffers to the active speaker buffers as speakers become &#x201c;active&#x201d; speakers. The x packets or m milliseconds of stored &#x201c;non-active&#x201d; speaker audio data can be used only up to a pre-determined jitter buffer fill-level in order to avoid introducing additional audio packet delivery delay to participants of the conference.

Efficient buffer allocation for current and predicted active speakers in voice conferencing systems

A method and computer program product allows for the efficient allocation of buffers (e.g., first-in first-out (FIFO) queues) for current and predicted active speakers in voice conferencing systems. The method and computer program product, implemented by a server hosting an audio conference for a plurality of speakers, minimizes the loss of audio data for speakers as they switch from &ldquo;non-active&rdquo; to &ldquo;active&rdquo; status. This is accomplished by employing a set of active speaker buffers and a set of predicted active speaker buffers. The predicted active speaker buffers maintain a collection of the most recent x packets or m milliseconds of &ldquo;non-active&rdquo; speaker audio data, and transfer a portion of the data from the predicted active speaker buffers to the active speaker buffers as speakers become &ldquo;active&rdquo; speakers. The x packets or m milliseconds of stored &ldquo;non-active&rdquo; speaker audio data can be used only up to a pre-determined jitter buffer fill-level in order to avoid introducing additional audio packet delivery delay to participants of the conference.

A system, method and computer program product which allows both mixing (e.g., PC-based) and non-mixing (e.g., phone-based) clients to participate in a single audio conference. The system includes a hybrid multi-point control unit (i.e., conferencing server) that performs mixing for phone-based clients and multiplexing for PC-based clients. The method and computer program product determines which clients have the capability to mix multiple audio streams and which do not. For those clients capable of mixing, the server multiplexes the packets of audio data received from each client on the active speakers list into a multiplexed stream. For those clients that are not capable of mixing, the server mixes the packets of audio data received from each client on the active speakers list into one combined packet.

Hybrid server architecture for mixing and non-mixing client conferencing

According to one aspect of the disclosure, the present invention provides methods and arrangements for the removal or reduction of divergence artifacts between a transmitting codec and a receiving codec. One of a number of implementations includes using a deblocking filter in an inverse transformer loop and selectively disabling the filter upon certain conditions.

Deblocking filter for encoder/decoder arrangement and method with divergence reduction

A video signal decoding arrangement conceals visible artifacts attributable to errors in a data stream. The arrangement includes a data processing arrangement that decodes the data stream into image data and that replaces data corrupted by the errors with replacement data generated as a function of a previous frame. The image data is optionally stored in first and second memories.

Video signal decoding arrangement and method for improved error concealment

According to one aspect of the disclosure, the present invention provides methods and arrangements for the removal or reduction of divergence artifacts between a transmitting codec and a receiving codec.

Growing community of inventors

San Jose, CA, United States of America

Katherine Wang Kwan