Abstract: An audio system has multiple loudspeaker devices to produce sound corresponding to different channels of a multi-channel audio signal such as a surround sound audio signal. The loudspeaker devices may have speech recognition capabilities. In response to a command spoken by a user, the loudspeaker devices automatically determine their positions and configure themselves to receive appropriate channels based on the positions. In order to determine the positions, a first of the loudspeaker devices analyzes sound representing the user command to determine the position of the first loudspeaker device relative to the user. The first loudspeaker also produces responsive speech indicating to the user that the loudspeaker devices have been or are being configured. The other loudspeaker devices analyze the sound representing the responsive speech to determine their positions relative to the first loudspeaker device and report their positions to the first loudspeaker device.

Abstract: A microphone biasing circuit comprises a first amplifier having an output connected to a first node and an input connected to a second node; and a first feedback path connected from the first node to the second node. The first feedback path comprises a microphone having a first terminal connected to the first node and a second terminal connected to a third node, the microphone being configured to provide a sensed voltage at the first node in response to sound, the third node having a first DC bias voltage; and a first capacitor connected between the third node and the second node.

Abstract: A device is provided which mixes and records two types of audio signals processed under standards different from each other; in particular, an audio signal of ASIO standard and an audio signal of WDM standard. An audio interface is connected to a computer, and an audio signal is input to the computer. A mixer module of the computer mixes an audio signal which is effect-processed by an ASIO application and an audio signal which is reproduced by a WDM application, outputs the mixed signal to the audio interface, supplies the mixed audio signal to the WDM application, and records the sound.

Abstract: A method for generating a virtual engine sound and a virtual engine sound generating system using the same involve determining a basic waveform of a virtual engine sound based on engine variables including an engine RPM, primarily correcting a basic level of the basic waveform based on environmental variables to determine a primarily corrected waveform, secondarily correcting the primarily corrected waveform based on reaction variables to determine a final waveform, and generating a virtual engine sound having the final waveform through a sound generation device.

Abstract: A wireless mobile training device has a microprocessor, a first switch, a microphone, and a speaker. The microprocessor generates a first control signal to induce the speaker to emit a first audible instruction requesting a training program identifier. The microphone generates a first input signal corresponding to a received audible training program identifier. The microprocessor generates a second control signal to induce the speaker to emit a second audible instruction associated with a predetermined training program name, if the received audible training program identifier corresponds to the predetermined training program identifier. The microprocessor determines a first time interval for completing a first user task associated with the second audible instruction based on second and third input signals from either the first switch or the microphone.

Abstract: A MEMS microphone includes a base structure and a piezoelectric resonator body having a first end and a second end. The first end is fixedly supported by the base structure and the second end is free such that the piezoelectric resonator is cantilevered from the base structure. The MEMS microphone further includes a first electrode operably connected to the piezoelectric resonator body and a second electrode operably connected to the piezoelectric resonator body. A controller includes at least one circuit operably connected to the first and second electrodes and configured to drive the piezoelectric resonator body at a shear resonance frequency of the piezoelectric resonator body and to detect a difference in the shear resonance frequency from a baseline resonance frequency resulting from a sound pressure.

Abstract: Techniques for optimizing a player based on the addition of a second player are disclosed. In an embodiment, when a first player no longer needs to play certain audio frequencies due to the addition of a second player, the gain of the first player is automatically increased as part of the setup process. In another embodiment, when a first player needs to play certain audio frequencies, for example due to the removal of a second player, the gain of the first player is automatically decreased. Many other embodiments are disclosed.

Abstract: A method and apparatus for processing object-based audio signals is provided. The apparatus receives a plurality of object-based audio signals. Each object-based audio signal of the object-based audio signals includes audio waveform data and object metadata associated with the audio waveform data. The object metadata includes at least one of a loudness parameter or a power parameter associated with the audio waveform data. The apparatus determines a loudness metric based on the received object-based audio signals and based on the at least one of the loudness parameter or the power parameter for each object-based audio signal of the received object-based audio signals. In one configuration, the apparatus renders the received object-based audio signals to a set of output signals based on the determined loudness metric. In another configuration, the apparatus transmits (e.g., broadcast, file delivery, or streaming) the received object-based audio signals based on the determined loudness metric.

Abstract: A device includes an input for receiving an audio signal, a speaker to convert the audio signal into an audible sound, and a memory for storing remediation instructions and detection instructions. The device further includes a processor coupled to the input, the speaker, and the memory. The processor is configured to process the audio signal according to the detection instructions and the remediation instructions to modulate amplitude of the audio signal based on the remediation instructions.

Abstract: An example method is performed by a media playback system comprising a plurality of audio drivers having a first radiation pattern. The method includes receiving data representing audio content, where each datum of the data indicates (i) a frequency and (ii) an amplitude corresponding to the frequency. The method further includes, for each audio driver of the plurality of audio drivers, determining a transfer function; processing each datum of the data based on (i) the frequency indicated by the given datum and (ii) the determined transfer function; and providing, to the given audio driver, a respective signal representing the data processed for the given audio driver, thereby causing the plurality of audio drivers to play back the audio content according to a second radiation pattern that is different from the first radiation pattern. An example media playback system and an example non-transitory computer-readable medium related to the example method is also disclosed herein.

Abstract: An audio renderer comprises a receiver (801) receiving input data comprising early part data indicative of an early part of a head related binaural transfer function; reverberation data indicative of a reverberation part of the transfer function; and a synchronization indication indicative of a time offset between the early part and the reverberation part. An early part circuit (803) generates an audio component by applying a binaural processing to an audio signal where the processing depends on the early part data. A reverberator (807) generates a second audio component by applying a reverberation processing to the audio signal where the reverberation processing depends on the reverberation data. A combiner (809) generates a signal of a binaural stereo signal by combining the two audio components. The relative timing of the audio components is adjusted based on the synchronization indication by a synchronizer (805) which specifically may be a delay.

Abstract: An audio system includes a processor including an input configured to: receive a baseband audio signal and modulate the baseband audio signal to create a modulated audio signal comprising audio signal frequency components in a first frequency range; clip the modulated audio signal to create a clipped, modulated audio signal the clipped modulated audio signal comprising the audio signal frequency components in the first range and further comprising distortion frequency components outside the first frequency range. The system can further be configured to filter the clipped, modulated audio signal to remove frequency components outside the first frequency to remove distortion components outside that frequency range.

Abstract: Systems and methods are presented for power supply noise cancellation in a microphone package. A first node on a MEMS microphone package is coupled to an inverted voltage signal of a power supply output. A summing node is connected to both the first node and a second node. The second node is coupled to a non-inverted voltage signal of the power supply output. Because the parasitic capacitance of the MEMS microphone package is substantially equivalent relative to both the first node and the second node, the summing node outputs a noise-reduced summed voltage signal. An adjustable trim module is coupled to the first node and configured to adjust a reduction of a power supply noise in the summed voltage signal from the summing node.

Abstract: A system for improving synchronization of an acoustic signal to a video display includes a hearing aid comprising a hearing loss processor configured for signal processing in accordance with a hearing loss of a user of the hearing aid, the hearing aid being configured for receiving a first audio signal for synchronous presentation to the user viewing the video display, the hearing aid being configured for generating a first acoustic signal to be presented to the user of the hearing aid, the first acoustic signal comprising at least a first part being generated in response to the first audio signal. The system also includes a delay unit configured for applying a delay, such that synchronization of the at least first part of the first acoustic signal to the video display is improved.

Abstract: One example discloses an apparatus for voice communication, including: a first wireless device including a first pressure sensor having a first acoustical profile and configured to capture a first set of acoustic energy within a time window; wherein the first wireless device includes a near-field magnetic induction (NFMI) signal input; wherein the first wireless device includes a processing element configured to: receive, through the NFMI signal input, a second set of acoustic energy captured by a second pressure sensor, having a second acoustical profile, within a second wireless device and within the time window; apply a signal enhancement technique to the first and second sets of acoustic energy based on the first and second acoustical profiles; and output an enhanced voice signal based on applying the signal enhancement.

Abstract: A thermal protecting device of a speaker is provided. The thermal protecting device includes an amplifier and a temperature detecting unit. An input end of the amplifier receives an audio reference signal, an output end of the amplifier provides an audio output signal to the speaker. The temperature detecting unit receives an audio input signal to provide the audio reference signal, and detects an operation temperature of the speaker to determine an amplitude of the audio reference signal. The amplitude is inversely proportional to the operation temperature.

Abstract: The derivation of personalized HRTFs for a human subject based on the anthropometric feature parameters of the human subject involves obtaining multiple anthropometric feature parameters and multiple HRTFs of multiple training subjects. Subsequently, multiple anthropometric feature parameters of a human subject are acquired. A representation of the statistical relationship between the plurality of anthropometric feature parameters of the human subject and a subset of the multiple anthropometric feature parameters belonging to the plurality of training subjects is determined. The representation of the statistical relationship is then applied to the multiple HRTFs of the plurality of training subjects to obtain a set of personalized HRTFs for the human subject.

Abstract: A fader level set for a fader of a channel is displayed at a channel level display section in a channel screen by, for example, a length of a bar, and a channel level indicating level of output from the channel is overlay-displayed by an indicator on the channel level display section at a position corresponding to the level to be displayed. It is thereby possible to compare the fader level set for the fader of the channel and the channel level indicating level of output from the channel by checking the channel level display section in the channel screen.

Abstract: A sound processing device includes a shift detection unit that detects a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and a correction unit that corrects a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point.

Abstract: An audio system having low latency includes a digital audio processor as well as sensor inputs coupled to the processor. The sensor inputs may be microphone inputs. The audio processor operates at the same frequency as the sensor inputs, which is typically much higher than an audio signal provided to the audio processor. In some aspects the audio processor operates as a noise cancellation processor and does not include an audio input.