Abstract: A LIDAR device can accurately calculate distances to objects in an environment by classifying a signal received from a sensor as being a particular type of signal (e.g., saturated or unsaturated) and selecting, based on the type of signal, a detector for processing the received signal from among multiple detectors. For example, the multiple detectors may include different programming and/or circuitry for determining a time delay of arrival (TDOA) between a time that a light pulse was emitted to a time that a pulse reflected off an object was received at a light sensor. The output of the selected detector may then be used to calculate a distance to the object from which the received signal was reflected.

Abstract: A device can accurately discriminate an active pulse from noise by setting a dynamic noise floor that adjusts according to environmental conditions. For example, the device may discriminate, as the active pulse, a light pulse emitted by a light emitter of the system and reflected off an object to a light sensor, from noise such as sunlight glare by determining a dynamic noise floor and identifying, as an active pulse, at least a portion of the received signal that exceeds the dynamic noise floor for a threshold number of samples. The dynamic noise floor may be determined, for example, using a moving average of the received signal and/or shifting or scaling the noise floor based on other properties of the return signal.

Abstract: A time delay of arrival (TDOA) between a time that a light pulse was emitted to a time that a pulse reflected off an object was received at a light sensor may be determined for saturated signals by using an edge of the saturated signal, rather than a peak of the signal, for the TDOA calculation. The edge of the saturated signal may be accurately estimated by fitting a first polynomial curve to data points of the saturated signal, defining an intermediate magnitude threshold based on the polynomial curve, fitting a second polynomial curve to data points near an intersection of the first polynomial curve and the intermediate threshold, and identifying an intersection of the second polynomial curve and the intermediate threshold as the rising edge of the saturated signal.

Abstract: Techniques for using acoustic notifications output via a multi-channel speaker configuration for pedestrian notification are described. Computing device(s) can determine a plurality of signals for dynamically emitting a plurality of sounds via a plurality of speakers associated with an acoustic array disposed on a vehicle. Each signal of the plurality of signals can be emitted from speaker(s) of the plurality of speakers that is/are associated with a particular channel of a plurality of channels. Furthermore, the computing device(s) can cause a portion of the plurality of speakers to dynamically emit the plurality of sounds so that the plurality of sounds are spatialized across the at least one surface of the vehicle. As a result, a pedestrian proximate the vehicle can localize the vehicle as a source of the acoustic notification and perceive a size, or other geometric characteristic, of the vehicle.

Abstract: Techniques for using beam-formed acoustic notifications for pedestrian notification are described. Computing device(s) can receive sensor data associated with an object in an environment of a vehicle. The computing device(s) can determine first data for emitting a first beam of acoustic energy via speaker(s) of an acoustic array associated with the vehicle, and second data for emitting a second beam of acoustic energy via speakers of the acoustic array. The computing device(s) can cause the speaker(s) to emit the first beam in a direction of the object at a first time and the second beam in the direction of the object at a second time. Directions of propagation of the first beam and the second beam are offset so that the object can localize the source the acoustic notification, thereby localizing the vehicle in the environment.

Abstract: Techniques are described for performing adaptive noise suppression to improve handling of both speech signals and music signals at least up to super wideband (SWB) bandwidths. The techniques include identifying a context or environment in which audio data is captured, and adaptively changing a level of noise suppression applied to the audio data prior to bandwidth compressing (e.g., encoding) based on the context. For a valid speech context, an audio pre-processor may set a first level of noise suppression that is relatively aggressive in order to suppress noise (including music) in the speech signals. For a valid music context, the audio pre-processor may set a second level of noise suppression that is less aggressive in order to leave the music signals undistorted. In this way, a vocoder at a transmitter side wireless communication device may properly encode both speech and music signals with minimal distortions.

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 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 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 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 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.

Abstract: A method includes determining, at a decoder of a first device, an offset value corresponding to an offset between a first particular packet and a second particular packet. The first device includes a de-jitter buffer. The method also includes transmitting the offset value to an encoder of a second device to enable the second device to send packets to the first device based on the offset value.

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: 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: The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold.