Qualcomm Incorporated

Sang-Uk Ryu

Dipanjan Sen

Jongwon Shin

Lae-Hoon Kim

Erik Visser

Samir Kumar Gupta

Eddie L T Choy

Yinyi Guo

Wade Lyle Heimbigner

Ricardo De Jesus Bernal Castillo

Andre Schevciw

Andre Gustavo P Schevciw

Joel Linsky

Pei Xiang

Sherif Galal

Hyun Jin Park

Dinesh Ramakrishnan

Hui-Ya Liao Nelson

Jeremy Patrick Toman

Suresh Devalapalli

Subbarao Surendra Chakkirala

Nidish Ramachandra Kamath

Earl Schreyer

Roy Benjamin Silverstein

Shreyas Srikanth Payal

Mofei Zhu

Amit Sharma

Andre Gustavo Schevciw

A method includes receiving first data associated with a first power amplifier and second data associated with a second power amplifier. The method also includes generating a first amplitude limiting signal having gain parameters that are based on the first data and the second data. The first data includes at least one of a temperature measurement associated with the first power amplifier, a supply voltage measurement associated with the first power amplifier, a load resistance associated with the first power amplifier, or a gain associated with the first power amplifier. The method further includes modifying an audio signal based at least in part on the first amplitude limiting signal to generate a first gain-adjusted audio signal. The method also includes providing a first output audio signal to the first power amplifier for amplification. The first output audio signal is based at least in part on the first gain-adjusted audio signal.

Power amplifier management

In general, disclosed is a device that includes one or more processors, coupled to the memory, configured to perform an energy analysis with respect to one or more audio objects, in the ambisonics domain, in the first time segment. The one or more processors are also configured to perform a similarity measure between the one or more audio objects, in the ambisonics domain, in the first time segment, and the one or more audio objects, in the ambisonics domain, in the second time segment. In addition, the one or more processors are configured to perform a reorder of the one or more audio objects, in the ambisonics domain, in the first time segment with the one or more audio objects, in the ambisonics domain, in the second time segment, to generate one or more reordered audio objects in the first time segment.

Reordering of foreground audio objects in the ambisonics domain

Aspects of the disclosure relate to using a display as a sound emitter and may relate to an electronic device including a display. In particular a vibration sensor such as an accelerometer is physically coupled to the display and senses display vibration to provide a high accuracy feedback loop with respect to representing actual audio output from the display. The electronic device includes an actuator physically coupled to the display and configured to cause vibration of the display in response to an audio signal. The electronic device further includes a vibration sensor physically coupled to the display and configured to output a vibration sensor signal proportional to the vibration of the display due to the actuator. The electronic device further includes a processor operably coupled to the vibration sensor. The processor is configured to adjust the audio signal based on the vibration sensor signal from the vibration sensor.

Feedback control for calibration of display as sound emitter

Aspects of the disclosure relate to using a display as a sound emitter and may relate to an electronic device including a display. In particular a vibration sensor such as an accelerometer is physically coupled to the display and senses display vibration to provide a high accuracy feedback loop with respect to representing actual audio output from the display. The electronic device includes an actuator physically coupled to the display and configured to cause vibration of the display in response to an audio signal. The electronic device further includes a vibration sensor physically coupled to the display and configured to output a vibration sensor signal proportional to the vibration of the display due to the actuator.

Feedback control for display as sound emitter

In general, techniques are described for compressing decomposed representations of a sound field. A device comprising one or more processors may be configured to perform the techniques. The one or more processors may be configured to obtain a bitstream comprising a compressed version of a spatial component of a sound field, the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients.

Compression of decomposed representations of a sound field

In general, techniques are described for identifying a codebook to be used when compressing spatial components of a sound field. A device comprising one or more processors may be configured to perform the techniques. The one or more processors may be configured to identify a Huffman codebook to use when compressing a spatial component of a plurality of spatial components based on an order of the spatial component relative to remaining ones of the plurality of spatial components, the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients.

Identifying codebooks to use when coding spatial components of a sound field

In general, techniques are described for determining quantization step sizes for compression of spatial components of a sound field. A device comprising one or more processors may be configured to perform the techniques. In other words, the one or more processors may be configured to determine a quantization step size to be used when compressing a spatial component of a sound field, where the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients.

Quantization step sizes for compression of spatial components of a sound field

A personalized (i.e., speaker-derivable) bandwidth extension is provided in which the model used for bandwidth extension is personalized (e.g., tailored) to each specific user. A training phase is performed to generate a bandwidth extension model that is personalized to a user. The model may be subsequently used in a bandwidth extension phase during a phone call involving the user. The bandwidth extension phase, using the personalized bandwidth extension model, will be activated when a higher band (e.g., wideband) is not available and the call is taking place on a lower band (e.g., narrowband).

Personalized bandwidth extension

A method for signal level matching by an electronic device is described. The method includes capturing a plurality of audio signals from a plurality of microphones. The method also includes determining a difference signal based on an inter-microphone subtraction. The difference signal includes multiple harmonics. The method also includes determining whether a harmonicity of the difference signal exceeds a harmonicity threshold. The method also includes preserving the harmonics to determine an envelope. The method further applies the envelope to a noise-suppressed signal.

Systems and methods for audio signal processing

An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Loudness maximization with constrained loudspeaker excursion

Decomposition of a multi-source signal using a basis function inventory and a sparse recovery technique is disclosed.

Decomposition of music signals using basis functions with time-evolution information

Encoder-assisted frame loss concealment (FLC) techniques for decoding audio signals are described. A decoder may discard an erroneous frame of an audio signal and may implement the encoder-assisted FLC techniques in order to accurately conceal the discarded frame based on neighboring frames and side-information transmitted from the encoder. The encoder-assisted FLC techniques include estimating magnitudes of frequency-domain data for the frame based on frequency-domain data of neighboring frames, and estimating signs of the frequency-domain data based on a subset of signs transmitted from the encoder as side-information. Frequency-domain data for a frame of an audio signal includes tonal components and noise components. Signs estimated from a random signal may be substantially accurate for the noise components of the frequency-domain data. However, to achieve highly accurate sign estimation for the tonal components, the encoder transmits signs for the tonal components of the frequency-domain data as side-information.

Encoder-assisted frame loss concealment techniques for audio coding

In general, techniques are described for performing majority vote error correction techniques. In operation, a communication device comprising a control unit implements the majority vote error correction techniques. The control unit includes a link management module to request a first retransmission of a first communication received over a wireless communication medium in response to detecting a first uncorrectable error in the first communication, requests a second retransmission of the first communication in response to detecting a second uncorrectable error in a second communication received in response to the first retransmission request and receives a third communication in response to the second retransmission. The control unit also includes a majority vote module to, in response to detecting a third uncorrectable error in the third communication, perform a bit-wise majority vote on corresponding bits of the first, second and third communications to generate an error-corrected communication.

Majority vote error correction

A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules.

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San Diego, CA, United States of America

Sang-Uk Ryu