Abstract: A method of compensating for jitter in a packet stream is described. The method comprises placing undecoded frames extracted from packets in the packet stream into a jitter buffer while decoding frames from the jitter buffer and placing the decoded frames into a sample buffer at a rate determined using an average playout delay. The average playout delay is the running average of the playout delay calculated for each packet as each packet becomes available. The playout delay for each packet is the sum of a sample buffer delay and a jitter buffer delay. As each packet is received, the average playout delay is adjusted based on a comparison of the playout delay associated with the received packet to the current average playout delay.

Abstract: Architecture for enhancing the compression (e.g., luma, chroma) of a video signal and improving the perceptual quality of the video compression schemes. The architecture operates to reshape the normal multimodal energy distribution of the input video signal to a new energy distribution. In the context of luma, the algorithm maps the black and white (or contrast) information of a picture to a new energy distribution. For example, the contrast can be enhanced in the middle range of the luma spectrum, thereby improving the contrast between a light foreground object and a dark background. At the same time, the algorithm reduces the bit-rate requirements at a particular quantization step size. The algorithm can be utilized also in post-processing to improve the quality of decoded video.

Abstract: A method of compensating for jitter in a packet stream is described. The method comprises placing undecoded frames extracted from packets in the packet stream into a jitter buffer while decoding frames from the jitter buffer and placing the decoded frames into a sample buffer at a rate determined using an average playout delay. The average playout delay is the running average of the playout delay calculated for each packet as each packet becomes available. The playout delay for each packet is the sum of a sample buffer delay and a jitter buffer delay. As each packet is received, the average playout delay is adjusted based on a comparison of the playout delay associated with the received packet to the current average playout delay.

Abstract: A method of compensating for jitter in a packet stream is described. The method comprises placing undecoded frames extracted from packets in the packet stream into a jitter buffer while decoding frames from the jitter buffer and placing the decoded frames into a sample buffer at a rate determined using an average playout delay. The average playout delay is the running average of the playout delay calculated for each packet as each packet becomes available. The playout delay for each packet is the sum of a sample buffer delay and a jitter buffer delay. As each packet is received, the average playout delay is adjusted based on a comparison of the playout delay associated with the received packet to the current average playout delay.

Abstract: Embodiments are configured to provide communication features, including providing channel condition estimates for a communication path, such as packet loss, jitter, and/or available bandwidth, but are not so limited. In an embodiment, a method uses aspects of in-band data packets to provide channel condition estimates. In one embodiment, a system includes a bandwidth estimation component that operates to classify payload packets as part of performing capacity estimation and available bandwidth estimation operations.

Abstract: Architecture for enabling a communications endpoint to quickly recover from a packet loss, reducing duration of a signal dropout. A communications component sends a downlink of dependency-structured signals, such as audio and/or video signals using compressed frames between key frames. A multipoint control component (MCC) is located between the communications component and multiple endpoints, and distributes the downlink to the multiple endpoints. A frame caching component caches a key frame of the downlink. If a key frame is lost at one of the endpoints, the endpoint sends a packet loss report to the frame caching component. The key frame is resent from the frame caching component to the endpoint in response to the key frame loss. In this way, the frame caching component can respond to specific frame loss situations on any of the endpoints, without interfering with the performance on the other endpoints.

Abstract: To reduce audio glitch rendering buffer of an audio application is pre-filled with natural sounding audio rather than zeros. For every frame of audio sent for rendering, the rendering buffer is also pre-filled or the signal is stretched in the buffer in anticipation of a glitch. If the glitch does not occur, then the stretched signal is overwritten and the end user does not notice it. If the glitch does occur, then the rendering buffer is already filled with a stretched version of the previous audio and may result in sound that is acceptable. After recovery from the glitch, any new data is smoothly merged into the fake audio that was generated before.

Abstract: Embodiments are configured to provide communication features, including providing channel condition estimates for a communication path, such as packet loss, jitter, and/or available bandwidth, but are not so limited. In an embodiment, a method uses aspects of in-band data packets to provide channel condition estimates. In one embodiment, a system includes a bandwidth estimation component that operates to classify payload packets as part of performing capacity estimation and available bandwidth estimation operations.

Abstract: An error correction system determines a level of error correction protection to apply to a frame of video data to be transmitted by a sending endpoint to a receiving endpoint based on the predicted impact of packet loss as well as the importance of the frame based on inter-frame dependencies, frame size, packet loss probability, historical packet loss pattern, central processing unit (CPU) load, and available network bandwidth. At the receiving endpoint, when packet loss is detected for a particular frame, the receiving endpoint will attempt to recover the frame using protection packets received along with the video data.

Abstract: Architecture for enabling a communications endpoint to quickly recover from a packet loss, reducing duration of a signal dropout. A communications component sends a downlink of dependency-structured signals, such as audio and/or video signals using compressed frames between key frames. A multipoint control component (MCC) is located between the communications component and multiple endpoints, and distributes the downlink to the multiple endpoints. A frame caching component caches a key frame of the downlink. If a key frame is lost at one of the endpoints, the endpoint sends a packet loss report to the frame caching component. The key frame is resent from the frame caching component to the endpoint in response to the key frame loss. In this way, the frame caching component can respond to specific frame loss situations on any of the endpoints, without interfering with the performance on the other endpoints.

Abstract: Techniques for deploying, maintaining and configuring complex hardware and software systems are provided. An abstract configuration of the system describes the system's desired state. Each component of the system declares the general form of the resources it requires and an abstract representation of the versions of the services it both requires and provides. A configuration process uses the abstract configuration of the system and the descriptions of each of the components in the system to validate that the system can operate effectively, ensures that each component is in the correct state and at the right version, and generates the necessary interconnections for the application components to interoperate with each other.

Abstract: Systems, methods, and/or techniques (“tools”) are described that sort speakers in a network-enabled conference. In some cases, this sorted list of speakers indicates which speaker is dominant. With this sorted list, a participant's communication device may provide context about the speakers. In some cases a participant's communication device has a display that presents real-time video of the speakers or other visual indicia, such as each or the most dominant speaker's name, picture, title, or location. These and other context about speakers may help participants better understand discussions in network-enabled conferences.

Abstract: Techniques for upgrading and/or downgrading a data resource deployed on a machine from one version to another version are provided. An application component that defines the data resource may provide an up/down tool for use in changing the data resource component from one version to another version. The up/down tool comprises an up/down process and one or more version conversion rules. The up/down process utilizes the provided version conversion rules to determine the ability of the up/down tool to deploy the data resource and to actually deploy the data resource.

Abstract: Techniques for determining a complete address of a test point of an application from a relative address of the test point and a configuration of a system are provided. An application component that defines the application includes a relative address of the test point. An abstract configuration of a system specifies a location where an instance of the application is to be deployed within the system. The complete address of the test point is then formulated using the relative address of the test point and the indication of the location where an instance of the application is to be deployed within the system. The complete address of the test point can then be used to perform a health check of the instance of the application.

Abstract: Architecture for enhancing the compression (e.g., luma, chroma) of a video signal and improving the perceptual quality of the video compression schemes. The architecture operates to reshape the normal multimodal energy distribution of the input video signal to a new energy distribution. In the context of luma, the algorithm maps the black and white (or contrast) information of a picture to a new energy distribution. For example, the contrast can be enhanced in the middle range of the luma spectrum, thereby improving the contrast between a light foreground object and a dark background. At the same time, the algorithm reduces the bit-rate requirements at a particular quantization step size. The algorithm can be utilized also in post-processing to improve the quality of decoded video.

Abstract: To reduce audio glitch rendering buffer of an audio application is pre-filled with natural sounding audio rather than zeros. For every frame of audio sent for rendering, the rendering buffer is also pre-filled or the signal is stretched in the buffer in anticipation of a glitch. If the glitch does not occur, then the stretched signal is overwritten and the end user does not notice it. If the glitch does occur, then the rendering buffer is already filled with a stretched version of the previous audio and may result in sound that is acceptable. After recovery from the glitch, any new data is smoothly merged into the fake audio that was generated before.

Abstract: An error correction system determines a level of error correction protection to apply to a frame of video data to be transmitted by a sending endpoint to a receiving endpoint based on the predicted impact of packet loss as well as the importance of the frame based on inter-frame dependencies, frame size, packet loss probability, historical packet loss pattern, central processing unit (CPU) load, and available network bandwidth. At the receiving endpoint, when packet loss is detected for a particular frame, the receiving endpoint will attempt to recover the frame using protection packets received along with the video data.

Abstract: A method of compensating for jitter in a packet stream is described. The method comprises placing undecoded frames extracted from packets in the packet stream into a jitter buffer while decoding frames from the jitter buffer and placing the decoded frames into a sample buffer at a rate determined using an average playout delay. The average playout delay is the running average of the playout delay calculated for each packet as each packet becomes available. The playout delay for each packet is the sum of a sample buffer delay and a jitter buffer delay. As each packet is received, the average playout delay is adjusted based on a comparison of the playout delay associated with the received packet to the current average playout delay.

Abstract: Systems, methods, and/or techniques (“tools”) are described that sort speakers in a network-enabled conference. In some cases, this sorted list of speakers indicates which speaker is dominant. With this sorted list, a participant's communication device may provide context about the speakers. In some cases a participant's communication device has a display that presents real-time video of the speakers or other visual indicia, such as each or the most dominant speaker's name, picture, title, or location. These and other context about speakers may help participants better understand discussions in network-enabled conferences.

Abstract: Techniques for upgrading and/or downgrading a data resource deployed on a machine from one version to another version are provided. An application component that defines the data resource may provide an up/down tool for use in changing the data resource component from one version to another version. The up/down tool comprises an up/down process and one or more version conversion rules. The up/down process utilizes the provided version conversion rules to determine the ability of the up/down tool to deploy the data resource and to actually deploy the data resource.