Google Inc.

Amazon Technologies, Inc.

Chad Georges Seguin

Michael Kai Morishita

Michael Asfaw

David C Breece, Iii

Anthony Michael Fadell

Philip Ryan Hilmes

Jun Yang

Simon Prakash

Shijing Xian

Leonardo Kusumo

James East

Michael Pate

Andrew Phillis

Colin Randall McEnroe

Moonseok Kim

Josef Osterneck

David C Breece Iii

Techniques, methods, systems, and other mechanisms for measuring the excursion of a speaker while being actively driven. Measuring excursion can involve attaching a flexible printed coil (FPC), including a sense coil, to the speaker, and monitoring an induced current as produced though the sense coil, and further detecting that violation of an excursion limit for the speaker may likely occur.

Inductive excursion sensing for audio transducers

A method for producing an audio signal from a panel audio loudspeaker in a mobile device includes receiving information about a status of the mobile device from one or more sensors of the mobile device. The method further includes driving, with a first drive signal, a first actuator coupled to a display panel of the panel audio loudspeaker at a first location and simultaneously driving, with a second drive signal, a second actuator coupled to the display panel at a second location different from the first location. The method also includes varying a relative timing of the first and second drive signals based on the information about the status of the mobile device to vary a direction of an audio signal and/or a source location of the audio signal on the display panel.

Dual panel audio actuators and mobile devices including the same

An example implementation may involve driving an audio output module of a wearable device with a first audio signal and then receiving, via at least one microphone of wearable device, a second audio signal comprising first ambient noise. The device may determine that the first ambient noise is indicative of user speech and responsively duck the first audio signal. While the first audio signal is ducked, the device may detect, in a subsequent portion of the second audio signal, second ambient noise, and determine that the second ambient noise is indicative of ambient speech. Responsive to the determination that the second ambient noise is indicative of ambient speech, the device may continue the ducking of the first audio signal.

Voice-based realtime audio attenuation

In general, the subject matter described in this disclosure can be embodied in methods, systems, and program products for analyzing, by a computing system, one or more signals transmitted to an audio transducer. The computing system determines, based on the analyzing the one or more signals, an inductance of electronic components affecting the one or more signals. The computing system modifies, based on using the determined inductance to determine that atypical displacement of the membrane of the audio transducer has occurred, the one or more signals to compensate for the atypical displacement of the membrane of the audio transducer.

Detecting and compensating for pressure deviations affecting audio transducers

Embodiments disclosed in the present disclosure relate to vibration transducers. Such a transducer includes an electromagnet having a conductive coil. The conductive coil is configured to be driven by an electrical input signal to generate magnetic fields. The transducer further includes a magnetic diaphragm that is configured to mechanically vibrate in response to the generated magnetic fields. Additionally, the transducer includes a pair of cantilevered arms formed from damping steel. The cantilevered arms couple the magnetic diaphragm to a frame. The magnetic diaphragm vibrates with respect to the frame when the electromagnet is driven by the electrical input signal. Additionally, the pair of flexible support arms are connected to opposing sides of the magnetic diaphragm.

Damping spring

The present disclosure relates to managing audio signals within a user's perceptible audio environment or soundstage. That is, a computing device may provide audio signals with a particular apparent source location within a user's soundstage. Initially, a first audio signal may be spatially processed so as to be perceivable in a first soundstage zone. In response to determining a high priority notification, the apparent source location of the first audio signal may be moved to a second soundstage zone and an audio signal associated with the notification may be spatially processed so as to be perceivable in the first soundstage zone. In response to determining user speech, the apparent source location of the first audio signal may be moved to a different soundstage zone.

Systems and methods for spatial audio adjustment

An example method may be performed by a wearable computing device (WCD) that includes a bone conduction transducer (BCT). The method includes providing, to the BCT, an input signal that represents audio content. The method further includes sensing a time-varying voltage of the input signal and a time-varying current of the input signal. The method further includes determining an operating mode based on the time-varying voltage and the time-varying current. The method further includes processing the audio content according to the determined operating mode and using the processed audio content to continue providing the input signal to the BCT.

Closed loop audio processing for bone conduction transducer

The present application describes signal enhancements to reduce bone conduction sensations of a wearable computing device and applications thereof. An example apparatus includes a wearable computing device comprising a bone conduction transducer (BCT) configured to receive and be driven by an audio signal, a processor, and a data storage comprising instructions executable by the processor to: (1) determine an input gain level of the audio signal at a frequency range; (2) compare the determined input gain level at the frequency range to a threshold gain level at the frequency range; (3) based on the comparison, apply a multi-band compressor (MBC) configured to process the audio signal to reduce gain in at least a portion of a mid-band frequency range of the audio signal; and (4) drive the BCT with the processed audio signal.

Signal enhancements for audio

A gain control system is responsive to a user gain setting and to an effective signal level to calculate and adaptively vary an output gain to avoid output clipping. The effective signal level is calculated by smoothing the energy of an input audio signal in accordance with selected attenuation factors. In times during which fast transient sounds are occurring, attenuation factors corresponding to relatively low levels of attenuation are selected. In times during which fast transient sounds are not detected, attenuation factors corresponding to relatively higher levels of attenuation are selected. The effective signal level is held at a constant level during periods of silence in order to avoid increasing the output gain during these periods. The output gain is calculated based on a comparison of the effective signal level to a compression threshold.

Adaptive gain control

A loudness learning and balancing system for one or more audio channels. The system learns user volume preferences from manual interventions, providing effective and robust loudness learning to provide a consistent and balanced sonic experience. The system also reduces and/or eliminates incorrect attenuation or over-amplification of quiet interludes in the audio material, thereby eliminating artifacts such as noise pumping or breathing. In addition, the system reduces low level noise in the audio signals while preserving a wide dynamic range at most listening levels.

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Campbell, CA, United States of America

Chad Georges Seguin