Abstract: A method, an apparatus, and a system for video communications include: transmitting, from a first apparatus using a network, a first video stream of a first user to a second apparatus of a second user, wherein the first user is in video communication with the second user; receiving, from the second apparatus using the network, a second video stream of the second user; determining, by a processor, a reaction of the second user to an area of interest in the first video stream using the second video stream; and updating, in response to the reaction of the second user to the area of interest in the first video stream, a parameter for encoding the area of interest in the first video stream at the first apparatus.

Abstract: A real-time communication software application for generating spatial audio with uniform reverberation in a real-time communication session is performed by an electronic communication device. The application is adapted to remove the reverberation of recorded speech signals from far-end participants by the dereverberation approach, render the direct sound parts by filtering the output signals by head-related transfer functions of desired directions, generate reverberant sound parts by convolving the output signals from with uniform room impulse responses or an artificial reverberator, and combine direct and reverberant sound components to generate spatialized speech signals. When speakers and listeners are located in two virtual conference rooms, the reverberation of the two rooms are coupled. The reverberant sound parts are then generated by convolving the output signals and coupled RIRs from the two rooms.

Abstract: An apparatus for communication between a sending application and a receiving application of a receiving apparatus includes a processor that is configured to establish a stream for transmitting data between the sending application and the receiving application; receive a first request from the sending application to transmit metadata to the receiving application; receive a second request from the sending application to transmit application data to the receiving application; responsive to a determination that a frame that includes the application data and the metadata has a size that is smaller than or equal to a maximum frame size, construct the frame to include the application data and the metadata; and transmit the frame in a packet to the receiving apparatus.

Abstract: A computer-implemented method for suppressing noise from audio signal uses both statistical noise estimation and neural network noise estimation to achieve more desirable noise reduction. The method is performed by a noise suppression computer software application running on an electronic device. The noise suppression computer software application first transforms the speech signal in time domain into frequency domain before determining a statistical noise estimate and a neural network noise estimate. The noise suppression computer software application merges the two noise estimates to derive a final noise estimate, and determines and refines a noise suppression filter. The filter is applied to the speech signal in frequency domain to obtain an enhanced signal. The enhanced signal is transformed back into time domain.

Abstract: A system and method for determining the performance of a video codec for real-time communication applications. A set of network conditions is used to simulate certain typical real-world network conditions. A coded video stream is transmitted from the transmitting end to the receiving end under these conditions. The end-to-end latency and received video fluency are measured along with a set of existing video quality measures by a video codec performance evaluation system including a network model and a video codec quality analyzer. The decodable frame ratio, latency and video fluency are used as the performance metrics for real-time communication video quality evaluation. The video codec performance evaluation system does not send any video data over a network when it determines the real-time communication quality of the video codec.

Abstract: A system and method for determining the performance of a real-time communication application, such as a video codec and transmission of the coded data. The method factors in burst data packet loss in determining a performance measure. The method uses a Hidden Markov Model on a set of data packets of a unit of data to determine the performance measure in the form a probability that all the data packets within the set are received. The measure is used to fine tune the system settings. The method causes the real-time communication application to adjust parameters of the application for improved transmission of data packets. The adjustment of a parameter can be an increase or a decrease.

Abstract: A method for image generation based on a Generative Adversarial Network (GAN) including a generator, a discriminator, and an encoder, wherein outputs of the generator are mapped, by the encoder, to a latent space adaptable to manipulate at least one characteristics of images generated by the GAN, the method including generating, by the encoder, a first encoding E(Y) of a target image Y and a second encoding E(G(Z)) of a generated image G(Z) corresponding to the target image Y, wherein the first and second encodings E(Y) and E(G(Z)) map Y and G(Z) to the latent space having a lower dimension than dimensionality of Y and G(Z), wherein the encoder is trained to minimize the differences between the first and second encodings E(Y) and E(G(Z)), and the generator is trained by using the first and second encodings E(Y) and E(G(Z)) as part of a loss function.

Abstract: A new real-time video communication system includes a set of electronic devices. Each device runs a specialized real-time video communication software application including a video quality module. The video quality module retrieves network connection statistic data from a network connection module, and video encoder statistic data from a video quality module. The video quality module uses the network connection statistic data and the video encoder statistic data to determine a first objective video quality measure without extensive mathematical operations. The video quality module also uses the first objective video quality measure, the network connection statistic data and the video encoder statistic data to determine a second objective quality measure without extensive mathematical operations.

Abstract: A method, an apparatus for real-time multimedia communications using a software-defined network (SDN) are provided. The method includes receiving, in a periodic manner, a path metric associated with a first service node in the SDN and a second service node in the SDN, wherein the path metric comprises at least one of: a load status of at least one of the first service node or the second service node, or a transmission metric between the first service node and the second service node; and in response to receiving the path metric, updating a cascade network topology comprising an optimal path for transmitting multimedia data between a first edge node and a second edge node.

Abstract: A computer-implemented method for generating spatial audio with uniform reverberation in a real-time communication session is performed by a real-time communication software application running on an electronic communication device. The method includes removing the reverberation of recorded speech signals from far-end participants by the dereverberation approach, rendering the direct sound parts by filtering the output signals by head-related transfer functions of desired directions, generating reverberant sound parts by convolving the output signals from with uniform room impulse responses or an artificial reverberator, combining direct and reverberant sound components to generate spatialized speech signals. When speakers and listeners are located in two virtual conference rooms, the reverberation of the two rooms are coupled. The reverberant sound parts are then generated by convolving the output signals and coupled RIRs from the two rooms.

Abstract: A transport framework for heterogeneous data streams includes session management module and a connection management module. The session management module is configured to receive a request to establish a first stream that is used for transmitting or receiving data, where the request includes an express indication as to whether the first stream is reliable or unreliable; construct a first data frame based on application data; handoff the first data frame to the connection management module; and maintain a record for the first data frame that includes whether the first data frame is successfully transmitted to the receiver. The connection management module is configured to receive the first data frame of the first stream from the session management module; receive a second frame from the session management module; encapsulate the first data frame and the second frame in a packet; and transmit the packet to the receiver using an unreliable protocol.

Abstract: An automatic SDK testing case generation system includes a computer software application for automatically generating optimized test cases. The automatic SDK testing case generation system categorizes candidate APIs presented by an SDK to be tested. The categories conform to an API call category sequence. The system iterates through API calling paths, which conform to the API call category sequence, to generate optimized subsets of test cases. The combination of the optimized subsets of test cases forms the final set of test cases for being used to test the SDK.

Abstract: A wireless access point software application, including multiple software modules, dynamically determines the interference level of each channel. The application determines that a new channel is better suited to serve an RTC/RTE application over the present channel based on the interference levels and a set of interference measure thresholds. The application automatically creates a new basic service set on the new channel and associates the RTC/RTE device running the RTC/RTE application with the new basic service set. The application further refuses to associate the RTC/RTE device with the previous channel when the interference level on the new channel remains low based on a second set of interference measure thresholds. In addition, the application allows additional RTC/RTE devices to join the new basic service set with the first RTC/RTE device when the interference level on the new channel remains low based on the second set of interference measure thresholds.

Abstract: Respective bandwidth information of respective connected devices of active media sessions are obtained. At least one additional media session is detected to have become active. A first total bandwidth of the at least one additional media session is determined. At least one scaling factor is determined using the respective bandwidth information and the first total bandwidth. The at least one scaling factor is transmitted to at least one connected device of the at least one of the respective connected devices. The at least one scaling factor is determined so as to keep a second total bandwidth of the active media sessions and the at least one additional media session below or equal an upper bandwidth limit of the communications network.

Abstract: A reference-order AL-FEC system for recovering network video data packet loss during real-time video communication includes a packetizer, a reference-order AL-FEC encoder, a reference-order AL-FEC decoder and a depacketizer. The packetizer constructs source symbols from source packets of a current frame. The encoder generates a repair symbol from the source symbols of the current frame and other reference frames based on the reference-order, not time-order, between the frames within an encoding window. The encoder also generates a repair packet based on the repair symbol. The decoder recovers a lost source symbol based on the source symbols of the frames of the encoding window and the repair symbol by decoding the repair packet. The decoding is achieved by solving a linear system of the repair symbol.

Abstract: A method of training a generator G of a Generative Adversarial Network (GAN) includes receiving, by an encoder E, a target data Y; receiving, by the encoder E, an output G(Z) of the generator G, where the generator G generates the output G(Z) in response to receiving a random sample Z and where a discriminator D of the GAN is trained to distinguish which of the G(Z) and the target data Y; training the encoder E to minimize a difference between a first latent space representation E(G(Z)) of the output G(Z) and a second latent space representation E(Y) of the target data Y, where the output G(Z) and the target data Y are input to the encoder E; and using the first latent space representation E(G(Z)) and the second latent space representation E(Y) to constrain the training of the generator G.

Abstract: The present invention relates to systems and methods for network labeling in order to enhance real time data transfers. A network for a real time data transfer is identified and predictive models for network performance are compared against to determine if the network is suitable for the data transfer. If so, then the real time data transfer may be completed as expected. However, if the network is predicted to be unsuitable for transmission an alternate means for connection may be suggested. The alternate suggestion may include delaying the data transfer until the network is expected to be in better conditions, connecting to another access point in the network, or switching to another network entirely. During the data transfer, the quality of the network is monitored in order to update the predictive models for the network's quality. Identifiers for the network may be utilized to keep track of the networks.

Abstract: A highly scalable automated testing system includes an automated testing controller, an automated testing message server and an automated testing message terminal running on a set of automated testing target devices. The controller and the terminals register themselves with the message server. A testing case is programmed for testing an SDK on a number of target devices concurrently and transformed into a set of command messages in JSON message format. The controller sends the set of messages to the message server. The message server then distributes the set of command messages to the terminals. In response, the terminal calls corresponding APIs of the SDK. The APIs called can be the same or different between the devices within the set of target devices. The SDK returns a result that is forwarded to the server. The server sends the results from the target devices to the controller. The controller verifies the results.

Abstract: A method for encoding multiple descriptions for a media stream includes: determining, for a current block of the media stream, a first number of duplicate blocks, determining, for the current block, a plurality of complementary forward gain factors, and determining a first number of descriptions for the current block by applying the plurality of complementary forward gain factors to the first number of duplicate blocks. A method for decoding multiple descriptions for a media stream includes: determining, for a current block of the media stream, a first number of descriptions from the media stream, in which each description comprises a scaled block corresponding to the current block, determining backward gain factors for the first number of scaled blocks, and determining the current block by applying the backward gain factors to the first number of scaled blocks.

Abstract: A method for recovering a current frame of an audio stream includes detecting that a current packet is lost, the current packet including an audio signal; splitting one or more frames into respective high-band signals and respective low-band signals, the one or more frames precede the current frame in the audio stream; inferring a current low-band signal of the current frame using, as inputs to a machine-learning model, the respective low-band signals; combining the inferred current low-band signal with the respective high-band signals to obtain the current frame; and adding the current frame to a playout buffer.