Abstract: An apparatus includes a first calculator configured to determine a long-term noise estimate of the audio signal. The apparatus also includes a second calculator configured to determine a formant-sharpening factor based on the determined long-term noise estimate. The apparatus includes a filter configured to filter a codebook vector to generate a filtered codebook vector. The filter is based on the determined formant-sharpening factor, and the codebook vector is based on information from the audio signal. The apparatus further includes an audio coder configured to generate a formant-sharpened low-band excitation signal based on the filtered codebook vector.

Abstract: A method and system for a reconfigurable robotic platform through a plurality of interchangeable service modules and adapted to engage in both autonomous and interactive maintenance and monitoring of a service area, the robotic platform configured to execute a stored service plan for the service area, the service plan comprising a service plan sequence comprising a service treatment step, alter the service plan sequence based on detecting an object that is a treatment obstacle in the service area as detected by the sensor, the treatment obstacle preventing execution of the service treatment step at a service treatment location within the service area, wherein the altered service plan sequence comprises moving around the treatment obstacle and skipping the service treatment step, store the service treatment location for later treatment, and resume execution of the service plan.

Abstract: A device includes a receiver configured to receive an audio frame of an audio stream. The device also includes a decoder configured to generate first decoded speech associated with the audio frame and to determine a count of audio frames classified as being associated with band limited content. The decoder is further configured to output second decoded speech based on the first decoded speech. The second decoded speech may be generated according to an output mode of the decoder. The output mode may be selected based at least in part on the count of audio frames.

Abstract: An apparatus includes a network interface configured to receive, via a circuit-switched network, a packet. The packet includes a primary coding of a first audio frame, redundant coding of a second audio frame, and one or more bits that indicate signaling information. The signaling information corresponds to a decode operation of at least one of the primary coding or the redundant coding. The apparatus further includes a decoder configured to decode a portion of the packet based on the signaling information.

Abstract: In a particular aspect, an apparatus includes a first network interface. The first network interface is configured to receive a packet via a packet-switched network. The packet may include a primary coding of a first audio frame and a redundant coding of a second audio frame. The apparatus further includes a processor. The processor is configured to generate a modified packet that includes one or more bits that indicate signaling information or packet decoding information. The signaling information or packet decoding information may correspond to decoding of at least one of the primary coding or the redundant coding. The apparatus further includes a second network interface configured to transmit the modified packet via a circuit-switched network.

Abstract: A device including gain shape circuitry configured to determine a number of sub-frames of multiple sub-frames that are saturated, the multiple sub-frames included in a frame of a high band audio signal. The device also includes gain frame circuitry configured to determine, based on the number of sub-frames that are saturated, a gain frame parameter corresponding to the frame.

Abstract: A device includes a decoder configured to receive an encoded audio signal at a decoder and to generate a synthesized signal based on the encoded audio signal. The device further includes a classifier configured to classify the synthesized signal based on at least one parameter determined from the encoded audio signal.

Abstract: A device includes a de-jitter buffer configured to receive a packet, the packet including first data and second data. The first data includes a partial copy of first frame data corresponding to a first frame of a sequence of frames. The second data corresponds to a second frame of the sequence of frames. The device also includes an analyzer configured to, in response to receiving the packet, generate a first frame receive timestamp associated with the first data. The analyzer is also configured to, in response to receiving the packet, generate a second frame receive timestamp associated with the second data. The first frame receive timestamp indicates a first time that is earlier than a second time indicated by the second frame receive timestamp.

Abstract: A device includes a first classifier and a second classifier coupled to the first classifier. The first classifier is configured to output first decision data that indicates a classification of an audio frame as a speech frame or a non-speech frame, the first decision data determined based on first probability data associated with a first likelihood of the audio frame being the speech frame and based on second probability data associated with a second likelihood of the audio frame being the non-speech frame. The second classifier is configured to output second decision data based on the first probability data, the second probability data, and the first decision data, the second decision data includes an indication of a selection of a particular encoder of multiple encoders available to encode the audio frame.

Abstract: A method for mitigating potential frame instability by an electronic device is described. The method includes obtaining a frame subsequent in time to an erased frame. The method also includes determining whether the frame is potentially unstable. The method further includes applying a substitute weighting value to generate a stable frame parameter if the frame is potentially unstable.

Abstract: A method includes determining an error condition during a bandwidth transition period of an encoded audio signal. The error condition corresponds to a second frame of the encoded audio signal, where the second frame sequentially follows a first frame in the encoded audio signal. The method also includes generating audio data corresponding to a first frequency band of the second frame based on audio data corresponding to the first frequency band of the first frame. The method further includes re-using a signal corresponding to a second frequency band of the first frame to synthesize audio data corresponding to the second frequency band of the second frame.

Abstract: A method and system for a reconfigurable robotic platform through a plurality of interchangeable service modules and adapted to engage in both autonomous and interactive maintenance and monitoring of a service area, the robotic platform configured to perform a wide variety of tasks utilizing the plurality of interchangeable service modules, and navigating through the service area utilizing a plurality of sensors and guided through a stored service plan for the service area.

Abstract: An apparatus includes a first calculator configured to determine a long-term noise estimate of the audio signal. The apparatus also includes a second calculator configured to determine a formant-sharpening factor based on the determined long-term noise estimate. The apparatus includes a filter configured to filter a codebook vector to generate a filtered codebook vector. The filter is based on the determined formant-sharpening factor, and the codebook vector is based on information from the audio signal. The apparatus further includes an audio coder configured to generate a formant-sharpened low-band excitation signal based on the filtered codebook vector.

Abstract: 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.

Abstract: A device includes a receiver, a buffer, a transmitter, and an analyzer. The receiver is configured to receive a plurality of packets that corresponds to at least a subset of a sequence of packets. Error correction data of a first packet of the plurality of packets includes a partial copy of a second packet of the plurality of packets. The analyzer is configured to determine whether a particular packet of the sequence is missing from the buffer, and to determine whether a partial copy of the particular packet is stored in the buffer. The analyzer is also configured to send, via the transmitter, a retransmit message to a second device based at least in part on determining that the buffer does not store the particular packet and that the buffer does not store the partial copy of the particular packet.

Abstract: A method of processing an audio signal includes determining an average signal-to-noise ratio for the audio signal over time. The method includes, based on the determined average signal-to-noise ratio, a formant-sharpening factor is determined. The method also includes applying a filter that is based on the determined formant-sharpening factor to a codebook vector that is based on information from the audio signal.

Abstract: A receiver is configured to receive packets that correspond to at least a subset of a sequence of packets and that include error correction data. The error correction data of a first packet of the packets includes a partial copy of a second packet. A buffer is configured to store the packets. An analyzer is configured to determine whether a first particular packet of the sequence is missing from the buffer, to determine whether a partial copy of the first particular packet is stored in the buffer as error correction data in a second particular packet, to update a value based at least in part on whether the first particular packet is missing from the buffer and the partial copy of the first particular packet is stored in the buffer, and to adjust an error recovery parameter based at least in part on the value.

Abstract: A particular method includes determining, based on spectral information corresponding to an audio signal that includes a low-band portion and a high-band portion, that the audio signal includes a component corresponding to an artifact-generating condition. The method also includes filtering the high-band portion of the audio signal and generating an encoded signal. Generating the encoded signal includes determining gain information based on a ratio of a first energy corresponding to filtered high-band output to a second energy corresponding to the low-band portion to reduce an audible effect of the artifact-generating condition.

Abstract: A particular method includes determining, at a device, a voicing classification of an input signal. The input signal corresponds to an audio signal. The method also includes controlling an amount of an envelope of a representation of the input signal based on the voicing classification. The method further includes modulating a white noise signal based on the controlled amount of the envelope. The method also includes generating a high band excitation signal based on the modulated white noise signal.

Abstract: A method for adjusting a delay of a buffer at a receiving terminal includes determining, at a processor, a partial frame recovery rate of lost frames at the receiving terminal. The method also includes adjusting the delay of the buffer based at least in part on the partial frame recovery rate.