Abstract: A method of providing custom audio processing in an audio device having an embedded audio DSP framework is disclosed. The audio device has an associated application that runs on a personal computing device. The method comprises receiving a notification of an availability of an audio processing configuration file, the audio processing configuration file identifying audio modules, interconnections between the audio modules and related settings for custom audio processing, transmitting a request for the audio processing configuration file from the personal computing device, receiving the audio processing configuration file at the personal computing device, transmitting the audio processing configuration file to the audio device from the personal computing device, and loading the audio processing configuration file into the audio device.

Abstract: A method of providing custom audio processing in an audio device having an embedded audio DSP framework is disclosed. The audio device has an associated application that runs on a personal computing device. The method comprises receiving a notification of the availability of an audio processing configuration file, the audio processing configuration file identifying audio modules, interconnections between the audio modules and related settings for custom audio processing, transmitting a request for the audio processing configuration file from the personal computing device, receiving the audio processing configuration file at the personal computing device, transmitting the audio processing configuration file to the audio device from the personal computing device, and loading the audio processing configuration file into the audio device.

Abstract: Audio data processing is done in multiple data processors in a multiprocessor system or in multiple processing instances in a multi-instance system. The system includes shared memory, the shared memory being read and write accessible to all of the data processors or instances. The multiprocessor or multi-instance audio processing is designed by placing representations of audio processing modules in a user interface, as well as a representation of an inter-process communications (IPC) module that includes a target core or target processing instance identifier associated therewith. The specified target core or target instance identifier is applied downstream from the representation of the IPC module such that the target core or instance identifier is associated with the representations of the subsequent downstream audio modules in the audio design.

Abstract: A method of providing audio processing in an audio device, executed by one or more processors, comprises receiving an audio signal, performing a subjective quality assessment on the audio signal to generate a subjective quality assessment score, based on the subjective quality assessment score, providing control parameters to a signal processing module, and processing the audio signal, by the signal processing module, based on the control parameters. The subjective quality assessment may be a Mean Opinion Score (MOS) performed by a trained machine-learning model.

Abstract: A method of providing custom audio processing in an audio device having an embedded audio DSP framework is disclosed. The audio device has an associated application that runs on a personal computing device. The method comprises receiving a notification of the availability of an audio processing configuration file, the audio processing configuration file identifying audio modules, interconnections between the audio modules and related settings for custom audio processing, transmitting a request for the audio processing configuration file from the personal computing device, receiving the audio processing configuration file at the personal computing device, transmitting the audio processing configuration file to the audio device from the personal computing device, and loading the audio processing configuration file into the audio device.

Abstract: A method of processing microphone signals generates a specified or required multichannel microphone input for a voice recognition system. The method comprises receiving first and second microphone signal from first and second microphones that have a spacing that is required by a voice response or recognition system. The first and second microphone signals are processed using at least one additional microphone signal to generate a first modified microphone signal and a second modified microphone signal respectively. The first and second modified microphone signals are then provided as required microphone inputs to a voice response or recognition system.

Abstract: Generating an audio processing design includes placing representations of audio processing modules using a user interface. The method includes causing the display of the representations of the audio processing modules in the user interface, receiving user input to place a representation of a start thread module in the audio processing design between a representation of an upstream audio module and representations of downstream audio modules, the representation of the start thread module having data associated therewith that defines a buffer up factor and a thread identifier, receiving user input to specify the buffer up factor and thread identifier associated with the representation of the start thread module, and applying the specified thread identifier and a sample block size that is derived from the specified buffer up factor, downstream from the representation of the thread start module.

Abstract: A method and apparatus for handling both echo cancellation and noise in a given space using a single adaptive filter. The proposed method utilizes just a single adaptive filter and without any control logic, the system is capable of handling both the AEC and the noise in a given space. The technology as proposed and claimed herein proposes using a single adaptive filter to cancel referenced noise AEC and unreferenced noise ANC in a given physical space using a beamformer. By adding the far end reference and a beam pointed dynamically on the noise source in the given physical space this can be achieved. This invention also reduces the processing load and memory.

Abstract: A method and system for improving communications conferencing systems that experience continuous double-talk where the communication includes an intended continuous or intermittent soundtrack or other intended continuous sound. The technology as disclosed and claimed herein uses several techniques to mask the residual echo and make it less audible.

Abstract: A method and apparatus for handling both echo cancellation and noise in a given space using a single adaptive filter. The proposed method utilizes just a single adaptive filter and without any control logic, the system is capable of handling both the AEC and the noise in a given space. The technology as proposed and claimed herein proposes using a single adaptive filter to cancel referenced noise AEC and unreferenced noise ANC in a given physical space using a beamformer. By adding the far end reference and a beam pointed dynamically on the noise source in the given physical space this can be achieved. This invention also reduces the processing load and memory.

Abstract: An apparatus and method for an audio user interface solution that avoids the time variation problems seen when using an acoustic echo canceler in an audio system. The solution as disclosed and claimed herein results in lower CPU usage in the audio system because a synchronous sample rate correction (SSRC) is performed on the sample rate multiple rate content and the asynchronous sample rate correction (ASRC) is applied on a small number of signals used to feed the ASRC.

Abstract: An apparatus and method for an audio user interface solution that avoids the time variation problems seen when using an acoustic echo canceler in an audio system. The solution as disclosed and claimed herein results in lower CPU usage in the audio system because a synchronous sample rate correction (SSRC) is performed on the sample rate multiple rate content and the asynchronous sample rate correction (ASRC) is applied on a small number of signals used to feed the ASRC.

Abstract: An audio system attachable to a computer includes a sound reproduction device for producing audible sound from audio signals. The sound reproduction device includes a radio tuner and a powered speaker. The audio system further includes a connector for connecting the sound reproduction device with a computer. The computer provides audio signals from a plurality of sources, the sources including a computer CD player, digitally encoded computer files stored on the computer, and a computer network connected to the computer. The sound reproduction device further includes control buttons for controlling at least one of the computer CD player, the digitally encoded computer files and the computer network.

Abstract: An audio system attachable to a computer includes a sound reproduction device for producing audible sound from audio signals. The sound reproduction device includes a radio tuner and a powered speaker. The audio system further includes a connector for connecting the sound reproduction device with a computer. The computer provides audio signals from a plurality of sources, the sources including a computer CD player, digitally encoded computer files stored on the computer, and a computer network connected to the computer. The sound reproduction device further includes control buttons for controlling at least one of the computer CD player, the digitally encoded computer files and the computer network.

Abstract: An audio system attachable to a computer includes a sound reproduction device for producing audible sound from audio signals. The sound reproduction device includes a radio tuner and a powered speaker. The audio system further includes a connector for connecting the sound reproduction device with a computer. The computer provides audio signals from a plurality of sources, the sources including a computer CD player, digitally encoded computer files stored on the computer, and a computer network connected to the computer. The sound reproduction device further includes control buttons for controlling at least one of the computer CD player, the digitally encoded computer files and the computer network.

Abstract: Methods and systems for processing a Global Positioning System (GPS) signal are provided. A method includes: transforming a time domain GPS signal to a frequency domain GPS signal; storing a frequency domain pseudorandom noise (PRN) signal; correlating the frequency domain GPS signal with the PRN signal at a plurality of frequencies, said correlating including: shifting the frequency domain GPS signal by an amount corresponding to one of the plurality of frequencies; downsampling the shifted frequency domain GPS signal; and multiplying the shifted frequency domain GPS signal by the stored frequency domain PRN signal to produce a correlated signal.

Abstract: Methods and devices for processing GPS signals are provided. The device includes: means for storing one or more lookup tables, each lookup table comprising a plurality of entries, each entry comprising an input segment and an output segment, wherein the output segment is a system response to the corresponding input segment; and processing means for receiving an input data sequence comprising one or more input segments, and for each of the input segments, retrieve from one of the lookup tables the output segments corresponding to the input segment, wherein the processing means is further configured to perform a time-shifted sum of one or more output segments to produce an output sequence that is a downsampled representation of the input data sequence.

Abstract: Methods and devices for processing GPS signals are provided. The device includes: a memory for storing one or more lookup tables, each lookup table including a plurality of entries, each entry including an input segment and an output segment, wherein the output segment is a system response to the corresponding input segment. The device further includes a processor configured to receive an input data sequence including one or more input segments, and for each of the input segments, retrieve from one of the lookup tables the output segments corresponding to the input segment. The processor is further configured to perform a time-shifted sum of one or more output segments to produce an output sequence that is a downsampled representation of the input data sequence.