Google Inc.

Telefonaktiebolaget Lm Ericsson (Publ)

Global IP Solutions, Inc.

Global IP Solutions (Gips) Ab

Global IP Solutions Ab

Henrik Lundin

Stefan Holmer

Serge Lachapelle

Mans Gustaf Sebastian Ullberg

Erik Kay

Jonas Erik Lindberg

Mikael Skoglund

Peter H&auml;ndel

Minyue Li

Alessio Bazzica

Niklas Blum

Patrik Westin

Mikael Pettersson

Lionel Koenig Gélas

Jesús De Vicente Peña

Tina Le Grand

Esbjörn Dominique

Michael Tschumi

Alexander Narest

A participant computing device connected to a teleconferencing session receives mixed audio data from a teleconference computing system orchestrating the session. The mixed audio data comprises a mix of audio captured by other devices connected to the session. The mixed audio and audio captured by the other devices prior to receipt of the mixed audio data are compared to determine that the participant computing device is co-located with at least one of the other devices. Information indicating that some of the other devices are featured devices that most actively contribute to the mixed audio is obtained. Co-location information for each of the featured devices is received. The co-location information is indicative of whether a featured device is co-located with the participant computing device. Based on the co-location information, a determination is made that one of the featured devices is co-located with the participant computing device.

Identifying co-located devices within a teleconferencing session

A first computing device captures audio data via an audio capture device. The audio data comprises probe signals from second computing devices located within the same area. The first computing device and the second computing devices are connected to a teleconference session. Based on the audio data, the first computing device generates a first encoding of the probe signals received. The first computing device receives a second encoding of probe signals received at a second computing device. The first computing device makes a determination that a degree of similarity between the first encoding and the second encoding is greater than or equal to a threshold degree of similarity. The first computing device generates co-location information identifying the second computing device as a co-located device that is located within the same area as the first computing device.

Local signaling of encoded audio for detection of co-located devices

Control logic determines when a network condition has changed by evaluating network condition data. Based on a detected network condition change, the control logic reorders an encoder parameter controller execution sequence of a plurality of encoder parameter controllers. The control logic configures the plurality of reordered encoder parameter controllers so that an encoder parameter determination from a prior encoder parameter controller is used as a decision input for a subsequent encoder parameter controller. An encoder encodes data, such as audio data, video data or other type of data using the generated encoder control parameters from the reordered plurality of encoder parameter controllers. A related method is also disclosed.

Method and device for providing pipeline reordering of encoder parameter controllers for an encoder

Computer-implemented techniques can include establishing, between first and second computing devices, an audio communication session via a network. The first computing device can receive and temporarily store audio packets corresponding to an audio input signal obtained by the second computing device. When the audio packets cause the buffer to fill beyond or at a rate exceeding a loss threshold indicative of a temporary loss of communication via the network, the techniques can decode the audio packets to obtain an audio output signal, determine whether the audio output signal includes speech, and when the audio output signal includes speech, remove a redundant portion of the audio output signal corresponding to a pitch period to obtain a modified audio output signal. The modified audio output signal can then be output via a speaker of the first computing device.

Audio compensation techniques for network outages

A computer-implemented technique can include establishing an audio communication session between first and second computing devices and obtaining, by the first computing device, an audio input signal using audio data captured by a microphone. The first computing device can analyze the audio input signal to detect a speech input by its first user and can determine a duration of a detection period from when the audio input signal was obtained until the analyzing has completed. The first computing device can then transmit, to the second computing device, (i) a portion of the audio input signal beginning at a start of the speech input and (ii) the detection period duration, wherein receipt of the portion of the audio input signal and the detection period duration causes the second computing device to accelerate playback of the portion of the audio input signal to compensate for the detection period duration.

Techniques for decreasing echo and transmission periods for audio communication sessions

Methods and systems are provided for removing transient noises (e.g., keyboard 'clicks') detected in an audio signal. An adaptive jitter buffer and packet loss concealer ('JB/PLC') is deployed on the transmitting side of a system (e.g., a Voice-over-IP (VoIP) system) to remove transient noises detected in an audio signal before the signal is transmitted over a network to the receiving side of the system. By utilizing the adaptation capabilities of the JB/PLC, the methods and systems provided produce better-sounding transient removal than other existing approaches.

Audio click removal using packet loss concealment

Disclosed herein is a method of transmitting a video bitstream to a first and at least a second endpoint with varying bandwidth capacities including identifying bandwidth capacities of the first and second endpoints, the second endpoint having less bandwidth capacity than the first endpoint, encoding at least a portion of the video bitstream to generate at least one version of a first data partition and a plurality of versions of at least a second data partition, the plurality of versions of the second data partition including at least one high quality version and at least one low quality version of the second data partition, transmitting the at least one version of the first partition and the at least one high quality version of the second partition to the first endpoint and transmitting the at least one low quality version of the second partition to the second endpoint.

System, method and apparatus for multipoint video transmission

A system, apparatus, and method for estimating available bandwidth for transmitting a media stream over a network, the media stream having a plurality of frames. One method includes receiving some of the plurality of frames, each frame of the plurality of frames having a inter-frame size differential and an inter-arrival time differential, detecting whether at least some of the inter-arrival time differentials are outside of a steady-state range using at least some of the inter-frame size differentials, and estimating an available bandwidth based on the detected change and using a processor.

Apparatus and method for media transmission bandwidth control using bandwidth estimation

A buffer level for jitter data buffer is determined. A frame payload size difference is determined for a plurality of video frames encoded into data packets sequentially received from a network. The difference is a difference in a payload size of a current frame and a previous frame. A frame network transit delay is determined as a difference in a transport time between the current frame and the previous frame and an expected transport time between the current frame and the previous frame. A slope and a variance of a linear relationship between the frame payload size difference and the frame network transit delay are determined for the plurality of video frames. Finally, a buffer level of a jitter data buffer is determined using a maximum frame payload size, an average frame payload size, the slope and the variance.

Method and device for determining a jitter buffer level

Disclosed herein is a method of decoding a sequence of prediction-coded video frames using a primary and a secondary state-dependent decoder. The method includes receiving an incomplete frame at the primary decoder, copying a state of the primary decoder into the secondary decoder, decoding and outputting the incomplete frame and subsequent frames at the primary decoder, receiving data restoring the incomplete frame, decoding the restored incomplete frame and subsequent frames at the secondary decoder until the secondary decoder has caught up with the primary decoder; and copying the state of the secondary decoder into the primary decoder; wherein the secondary decoder has access to the subsequent frames received after the incomplete frame.

Loss-robust video transmission using plural decoders

A system, method and apparatus for transmitting a packetized video stream over a network comprising receiving a frame of a video stream having a plurality of partitions of varying sizes, the plurality of partitions having an ordered sequence, identifying one or more of the plurality of partitions having a size that is less than a predetermined maximum, and allocating the identified one or more of the plurality of partitions into a plurality of packets in a manner that: (a) results in each packet having a size that is less than the predetermined maximum; (b) minimizes a cost value that is based at least in part on the difference between the size of the smallest packet and the size of the largest packet; and (c) maintains the allocated partitions in the ordered sequence. After allocation, the packets are transmitted over the network.

Method to packetize an encoded video frame

It is disclosed a method for determining a buffer level of a jitter data buffer () comprised in a receiver () adapted to sequentially receive data packets from a communications network (), wherein frames are encoded into the data packets. Each frame may comprise timestamp information and payload size information. The buffer level is determined on the basis of a first part of frame arrival delay related to payload size variation between frames and a second part related to the amount of crosstraffic in the communications network (). It is further disclosed a receiver () adapted to sequentially receive data packets from a communications network (), wherein frames are encoded into the data packets, the receiver () comprising a data jitter buffer () and being adapted to perform the method.

The invention relates to a method and a receiver having control logic means for determining a target packet level of a jitter buffer adapted to receive packets with digitized signal samples, which packets are subject to delay jitter, from a packet data network. According to the invention, the jitter buffer is made adaptive to current network conditions, i.e., the nature and magnitude of the jitter observed by the receiver, by collecting statistical measures that describe these conditions. The target buffer level is determined with regard to the effect of packet losses in terms of duration of the discontinued playback of the true signal. This effect is derived from statistical measures of the network conditions as perceived by the receiving side and as reflected by a probability mass function which is continuously updated with packet inter-arrival times. The target buffer level is the result of minimization of a cost function which weights the internal buffer delay and an expected length of buffer underflow.

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Sollentuna, Sweden

Henrik Lundin