Qualcomm Incorporated

Qualcomm Technologies International, Ltd.

Cambridge Silicon Radio Limited

Richard Turner

Laurent Wojcieszak

Justin Hundt

Joel Linsky

Megan Lucy Taggart

Gary Sands

Olaf Josef Hirsch

Ahmed Ragab Elsherif

Benjamin Campbell

Eric Y Tsou

Raghavendra Bhat Noojady Krishna

Bharath Kumar Tirunagaru

Nachiket Shankar Deshpande

John Oliver

Derrick Rea

This disclosure provides methods, components, devices and systems for low-latency parameter updates. In some aspects, a wireless communication device, such as a handset or an access point (AP), may transmit, to one or more wireless audio devices, an indication of a set of updated parameters associated with an extended personal area network (XPAN) of the wireless communication device. The wireless communication device may embed the indication of the set of updated parameters via a real-time protocol (RTP) header or in a padding section of a payload data of an audio data packet. A wireless audio device may extract the set of updated parameters from the audio data packet and may transmit an acknowledgement (ACK) to the wireless communication device. As a result of receiving an ACK from the wireless audio device(s), the wireless communication device may communicate with the wireless audio device(s) in accordance with the updated parameters.

Low-latency parameter updates for extended personal area networks

Methods, systems, and devices for wireless communications are described. In a wireless audio system, a first wireless device may generate a first media stream associated with a first media mode and a second media stream associated with a second media mode, buffer the first media stream in accordance with a first latency time value for adjusting a first latency time of the first media stream and buffering the second audio stream, independently from the first media stream, in accordance with a second latency time value for adjusting a second latency time of the second media stream, and transmit a mixed media stream including the buffered first media stream mixed with the buffered second media stream to a second wireless device.

Seamless transitions between media modes

Methods, devices, non-transitory computer-readable medium, and systems are described for compressing audio data. The techniques involve obtaining a sequence of digitized samples of an audio signal, performing a transform using the sequence of digitized samples, to generate a plurality of spectral lines, obtaining a group of spectral lines from the plurality of spectral lines, and quantizing the group of spectral lines to generate a group of quantized values. Quantizing the group of spectral lines to generate the group of quantized values may comprise performing a specialized rounding operation on a spectral line selected from the group of spectral lines and using the specialized rounding operation to force a group parity value, computed for the group of quantized values, to a predetermined parity value. One or more data frames based on the group of quantized values may be outputted.

Stream conformant bit error resilience

A device includes one or more processors configured to execute instructions to obtain, at a first audio output device, first spatial audio data and a first reference time associated, and to cause the first reference time and data representing at least a portion of the first spatial audio data to be transmitted from the first audio output device. The instructions further cause the one or more processors to receive, at the first audio output device from a second audio output device, second spatial audio data and a second reference time. The instructions further cause the one or more processors to, based on the first reference time and the second reference time, time-align the first spatial audio data and the second spatial audio data to generate combined audio data representing a three-dimensional (3D) sound field and to generate audio output based on the combined audio data.

Spatial audio data exchange

In general, various aspects of techniques are described to enable application directed latency control for wireless audio streaming. A source device comprising a memory and a processor may perform the techniques. The memory may store at least a portion of audio data. The processor may execute an application that outputs the audio data, and a request for a quality of service concerning audio processing applied to the audio data. The processor may determine whether the source device is currently displaying the application, and configure, responsive to the determination that the source device is currently displaying the application, a wireless audio processing path to achieve the requested quality of service. The processor may next process, by the wireless audio processing path, the audio data to obtain a bitstream representative of the audio data.

Application directed latency control for wireless audio streaming

Examples herein include apparatus and methods for overriding a media access control module acknowledgment and non-acknowledgment scheme. In one example, a codec module determines if an error-corrected packet is acceptable in response to the detection by the media access module of an uncorrectable error in a received packet, and overrides the media access control module to prevent a non-acknowledgment response from being sent when the error-corrected packet is acceptable.

Media access controller with a codec error model

Methods, systems, and devices for wireless communications are described. The described techniques provide for generating, by an encoding device, one or more entropy symbols, the length of which changes responsive to errors, stacking the entropy symbols into fixed intervals, and selecting a parity bit. The encoding device may divide entropy symbols that are longer than the fixed interval duration and stack the excess portions of the long entropy symbols with shorter entropy symbols in other intervals. The encoding device may transmit the slacked data packet according to the stacking. A decoding device may receive the data packet, identify the locations of the entropy symbols and the selected parity bit, check the parity of each entropy symbol, identify error bits based on the locations of the entropy symbols within multiple fixed intervals, and may correct error bits based on the stacked intervals, the parity bit, and an error mask.

Quantization codeword selection for low cost parity checking

A source device comprising a memory and a processor may be configured to perform techniques described in this disclosure. The memory may store at least a portion of the audio data. The processor may obtain, from a compressed version of the audio data, a symbol, and obtain a plurality of intervals, each having a same bit length. The processor may obtain a portion of the symbol within the bit length and an excess portion of the symbol over the bit length, and specify, in a first interval, the portion of the symbol. The processor may also specify, in a second interval, the excess portion of the symbol, and apply, to the first interval and the second interval, error resiliency. The processor may specify, in a bitstream representative of the compressed version of the audio data, the first error resilient interval and the second error resilient interval.

Error resiliency for entropy coded audio data

A first device may establish, with a second device, a short-range wireless communications link. The first device may determine a time interval between each of a set of packets to be transmitted over a short-range wireless communications link with a second device. The first device may encode data with a bitrate that is based on the time interval. The first device may packetize the encoded data in the set of packets. The first device may transmit the set of packets to the second device or the short-range wireless communications link, and each of the set of packets may be separated by the time interval when transmitted.

Low-latency audio streaming with communication coexistence

A system for distributing audio or video data across a network, wherein an input data rate controller at a server controls the transmission rate to a client such that it matches the intended rendering rate of the data, and time stamps the data using a clock that is synchronized with the clock of the client. The client monitors the rate at which data is received, compares the rate against the rate at which the data is actually rendered, and adjusts the rate at which the data is actually rendered depending on the comparison.

Reduced latency media distribution system

A method of calculating a transmission delay value for data transmitted from a first device to a second device across a communications link. The method involves calculating a plurality of measured values representing the transmission delay measured by the second device over time; detecting a minimum boundary of the measured values; and deriving the transmission delay value from the minimum boundary. The transmission delay value may be used to synchronize the clock signal of the second device to the clock signal of the first device.

Method and apparatus for calculating transmission delay across a network

A system for distributing audio or video data across a network, wherein an input data rate controller at a server controls the transmission rate to a client such that it matches the intended rendering rate of the data, and time stamps the data using a clock that is synchronized with the clock of the client. The client uses the time stamps and its synchronized clock signal to control the rendering of the data.

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Belfast, United Kingdom

Richard Turner