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 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 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.

Abstract: A new real-time video communication device improves the clarity of shared screen by automatically adjusting the key frame's quantization parameter values. The adjustment depends on the underlying network's connection condition. When the connection bandwidth deteriorates, the key frame's quantization parameter value is further decreased on the value output from a standard H.264 encoder to derive a desirable quantization parameter value and thus achieve a desirable user experience of the shared screen. Moreover, the new real-time video communication device adjusts the frame rate of the shared screen based on the present quantization parameter value to achieve a more desirable user experience of the screen sharing.

Abstract: A new real-time spatial audio rendering system includes a real-time spatial audio rendering computer software application adapted to run on a communication device. The application renders stereo audio from mono audio sources in a virtual room of a listener. The listener can be mobile. The stereo audio is rendered for each listener within the room. The real-time spatial audio rendering system has two different modes, with and without reverberation. Reverberation can provide the sense of the dimensions of the room, First, the anechoic processing module produces the anechoic stereo audio that provides the sense of direction and distance of spatial audio. When reverberation is desired, the reverberation processing module is also performed to provide the sense of the room's dimensions by the spatial audio.

Abstract: A computer software application running on a wireless communication device determines whether an application scenario is urgent or nonurgent, and determines whether the user state is interest, uninterested or absent. The application sends the application scenario and the user state to a wireless networking device. The wireless networking device determines the user roles of different wireless communication devices. It further adjusts priorities of data packets destined to the wireless communication device based on the application scenario, the user state and the user role when downlink to the wireless communication device is congested. The priority is decreased when the application scenario is unurgent, the user role is a listener, and the user state is uninterested or absent.

Abstract: A method for determining and providing upstream directives in real-time communication over a wireless network is performed by an RTC/RTE application running on a mobile device and an access point application running on a Wi-Fi access point. The access point application determines the values of a set of upstream Wi-Fi signal quality parameters including at least one of an upstream RSSI parameter, an upstream SNR parameter, an upstream link speed parameter, and an upstream remaining bandwidth parameter of the mobile device. Each value is compared to a set of upstream thresholds to derive an upstream Wi-Fi connection quality measure. Each set of upstream thresholds includes at least two different values. The RTC/RTE application determines an upstream directive from the upstream Wi-Fi connection quality measure, and presents the upstream directive to a user in an audio or video form.

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 new audio quality assessment system includes an assessment system running in a receiver system of a VoIP communication system. The new audio quality assessment system determines an accurate MOS of a VoIP call within a time window. The audio quality assessment system determines an effective PLC counter, a PLC impact factor, an effective AS counter, an AS impact factor, a network impact factor, a codec type of the received voice packets, a bitrate of the received voice packets, an initial MOS from a configured codec-bitrate MOS table, and determines the accurate MOS based on these data. The determined MOS is more accurate and efficiently obtained since it is based on efficiently collected statistics of the receiver system's modules and a pre-configured codec-bitrate MOS table.

Abstract: A wireless access point software application, adapted to be run within a wireless access point, classifies network data packets with classification identifiers provided by RTC/RTE devices without scanning the content of them. The wireless access point software application receives inbound and outbound packet classification identifiers from the RTC/RTE applications running on the RTC/RTE devices. The identifiers are provided to a hardware data packet classifier. The hardware data packet classifier applies the identifiers against ingress packets and egress packets respectively. Data packets of the same class are assigned with a same priority. The prioritized network data packets are scheduled for transmission based on their respective priorities. The scheduled network data packets are forwarded to respective network interfaces for transmission.

Abstract: An improved SDK includes a set of APIs and a crash handler registered with the operating system. Each API is an interface accessible by a computer software application. Up on entrance, each API determines the current thread identifier, and inserts it into a list if it is not already in the list. Each thread identifier corresponds to an API call counter, which is incremented by one at the entrance and decremented by one at the exit point of the API. The SDK also records the identifier of the thread it creates for callback functions. When a crash occurs, the crash handler is executed. It determines that the crash is related to a callback interface if the crash thread identifier matches the callback thread identifier. The crash is determined to be caused by the SDK if the API call counter corresponding to the crash thread identifier is greater than zero.