Qualcomm Incorporated

Vivek Rajendran

Venkatraman Atti

Venkatesh Krishnan

Subasingha Shaminda Subasingha

Daniel Jared Sinder

Ananthapadmanabhan Arasanipalai Kandhadai

Venkata Subrahmanyam Chandra Sekhar Chebiyyam

Stephane Pierre Villette

Ethan Robert Duni

Duminda Dewasurendra

Alberto Rico Alvarino

Nikolai Konrad Leung

Umesh Phuyal

Toru Uchino

Pravin Kumar Ramadas

Xiao Feng Wang

Bharat Shrestha

Min Wang

Sandip Sarkar

Chandra Mouli Polisetty

Alireza Heidari

Liangping Ma

Saurabha Rangrao Tavildar

Jeremy Patrick Toman

Dai Yang

Guillaume Konrad Sautière

Ashish Kumar Tawari

Zisis Iason Skordilis

Reza Barazideh

Venkatraman S Atti

Guillaume Konrad SAUTIÈRE

Robert Ethan Duni

Kumar Ashish Tawari

Venkatraman Srinivasa Atti

A method for wireless communication by a wireless device includes receiving a first bit sequence and a second bit sequence from an upper layer of a wireless communication protocol stack. The first bit sequence is assigned a first class and the second bit sequence is assigned a second class. The second bit sequence is included in a same medium access control (MAC) packet data unit (PDU) as the first bit sequence. The method also includes processing, at a physical layer, the first bit sequence with a first quality of service (QoS) level, and the second bit sequence with a second QoS level that is different from the first QoS level.

Radio access network (RAN) enhancement for unequal error protection (UEP)

Various aspects of the present disclosure generally relate to wireless communication. In some aspects, a network node may obtain an erroneous packet from a first communication device. The network node may output an indication of bit error rate information, based on the erroneous packet, to a second communication device to facilitate error mitigation at the second communication device. Numerous other aspects are described.

Using partially decoded packets for error mitigation at a voice decoder

A method, apparatus and computer-readable medium for transmitting a medium access control (MAC) protocol data unit (PDU) including one or more MAC service data units (SDUs) of a logical channel; and transmitting a first type of MAC subheader including or excluding first logical channel identifier (LCID) information and including or excluding first length information, wherein a first size of the first LCID information is equal to or smaller than a second size of second LCID information of a second type of MAC subheader that is different than the first type of MAC subheader, and a third size of the first length information is equal to or smaller than a fourth size of second length information of the second type of MAC subheader.

Protocol overhead reduction for medium access control

A receiver receives a first message including a first PDCP SN. The receiver may receive a second message including a second PDCP SN. The receiver may derive a second RTP SN based on at least one of a first RTP SN of the first message, the first PDCP SN, the second PDCP SN, or an RTP SN field of a second PDCP header composing the second message.

Protocol overhead reduction for packet data convergence protocol

A method for managing audio during a conference includes steering, at a spatial steering processor of a mobile device, first decoded audio to be projected from a speaker at a first angle. The method also includes steering, at the spatial steering processor, second decoded audio to be projected from the speaker at a second angle. The first decoded audio corresponds to a decoded version of a first audio stream from a first device associated with a first participant of the conference, and the second decoded audio corresponds to a decoded version of a second audio stream from a second device associated with a second participant of the conference. The first decoded audio is synchronized with the second decoded audio.

Conference audio management

Techniques are described for coding audio signals. For example, using a neural network, a residual signal is generated for a sample of an audio signal based on inputs to the neural network. The residual signal is configured to excite a long-term prediction filter and/or a short-term prediction filter. Using the long-term prediction filter and/or the short-term prediction filter, a sample of a reconstructed audio signal is determined. The sample of the reconstructed audio signal is determined based on the residual signal generated using the neural network for the sample of the audio signal.

Artificial intelligence based audio coding

A method of capturing video includes instructing a microphone to initiate an audio scan of different areas of interest to detect audio activity. The method also includes receiving a feedback signal from the microphone in response to initiating the audio scan. The feedback signal indicates audio activity in a particular area of interest. The method further includes adjusting targeting characteristics of a device based on the feedback signal to focus the device on the particular area of interest.

Device adjustment based on laser microphone feedback

A device includes a receiver configured to receive an audio frame of an audio stream. The audio frame includes information that indicates a coded bandwidth of the audio frame. The device also includes a decoder configured to generate first decoded speech associated with the audio frame and to determine an output mode of the decoder based at least in part on the information that indicates the coded bandwidth. A bandwidth mode indicated by the output mode of the decoder is different than a bandwidth mode indicated by the information that indicates the coded bandwidth. The decoder is further configured to output second decoded speech based on the first decoded speech. The second decoded speech is generated according to an output mode of the decoder.

Audio bandwidth selection

A method and an apparatus for estimating speech signal in split-domain is disclosed. The method includes performing LP analysis on a noisy speech signal to generate a first plurality of LPC and a first residual signal. The method also includes estimating speech LPC spectrum to generate cleaned LPC. The method further includes estimating speech residual spectrum to generate cleaned residual signal. The method also includes synthesizing output signals based on the cleaned LPC and the cleaned residual signal.

Split-domain speech signal enhancement

A method includes receiving a second audio frame at a decoder. The second audio frame follows a first audio frame in an audio signal and includes a first number of bits allocated to primary coding information associated with the second audio frame, a second number of bits allocated to redundant coding information associated with the first audio frame, and an indicator of a frame type of the first audio frame. In response to a frame erasure condition associated with the first audio frame, the second number of bits is determined based on the indicator and used to decode the first audio frame. In clean channel conditions, the first audio frame is received and decoded based on primary coding bits in the first audio frame, and the first number of bits is determined based on the indicator and used to decode the second audio frame.

Systems and methods of communicating redundant frame information

The disclosure generally relates to various methods to increase network coverage with respect to Voice-over-Internet Protocol (VoIP) sessions and/or other voice-based multimedia services. In particular, when establishing a voice session, two or more terminals may perform a codec negotiation to exchange information related to supported multimedia codecs and/or codec modes, jitter buffer management (JBM), and packet loss concealment (PLC) capabilities. Based on the exchanged information, a message may be sent to a base station to indicate the maximum packet loss rate (PLR) for each terminal. Additional techniques may ensure that the terminals use the most robust codecs or codec modes that are available when nearing the edge of coverage to help avoid unnecessary and/or excessive handovers to different radio access technologies.

Methods for increasing VoIP network coverage

A method of capturing audio includes initiating capture, at a laser microphone, of first audio of an area of interest. The first audio is captured while the laser microphone is focused on a first target surface associated with the area of interest. The method also includes generating adjustment parameters based on a feedback signal to adjust targeting characteristics of the laser microphone. The method further includes adjusting the targeting characteristics of the laser microphone based on the adjustment parameters to focus the laser microphone on a second target surface associated with the area of interest. The method also includes initiating capture, at the laser microphone, of second audio of the area of interest in response to adjusting the targeting characteristics. The second audio has an audio quality that is greater than the first audio.

Adjustable laser microphone

A method of controlling frame transmissions includes determining, at a first device, a frame erasure rate for a communication session between the first device and at least a second device. The method also includes comparing the frame erasure rate to an erasure threshold. The method further includes discarding an active speech frame if the frame erasure rate satisfies the erasure threshold.

Power gains and capacity gains for a relaxed frame erasure rate

A method includes extracting a voicing classification parameter of an audio signal and determining a filter coefficient of a low pass filter based on the voicing classification parameter. The method also includes filtering a low-band portion of the audio signal to generate a low-band audio signal and controlling an amplitude of a temporal envelope of the low-band audio signal based on the filter coefficient. The method also includes modulating a white noise signal based on the amplitude of the temporal envelope to generate a modulated white noise signal and scaling the modulated white noise signal based on a noise gain to generate a scaled modulated white noise signal. The method also includes mixing a scaled version of the low-band audio signal with the scaled modulated white noise signal to generate a high-band excitation signal that is used to generate a decoded version of the audio signal.

High band excitation signal generation

A method for managing audio during a conference includes receiving, at a first buffer of a mobile device, a first audio stream from a first device associated with a first participant of the conference. The method also includes receiving, at a second buffer of the mobile device, a second audio stream from a second device associated with a second participant of the conference. The method further includes generating a control signal at a delay controller of the mobile device. The control signal is provided to the first buffer and to the second buffer to synchronize first buffered audio that is output from the first buffer with second buffered audio that is output from the second buffer.

Growing community of inventors

San Diego, CA, United States of America

Vivek Rajendran