A portable device includes a controller that is responsive to a remaining power capacity of the battery, and a power consumption level of the portable device and based on user prioritized functional processing capability features, dynamically controls functional processing capability features of the device. The controller provides power for a higher priority feature at the expense of a lower priority functional processing capability feature consistent with the user prioritized functional processing capability features. A wireless portable device is also disclosed that includes a wireless signal strength determinator that determines a received signal strength of the wireless device and a controller that adjusts the functional processing capability feature of the wireless device based on the determined received signal strength and based on battery capacity information. In this way, received signal strength information is used as a further input to determine how to best adjust functional processing capability features of the device.

Portable device with priority based power savings control and method thereof

Nodes in a network include a pseudo-timestamp in messages or packets, derived from local pseudo-time clocks. When a packet is received, a first time is determined representing when the packet was sent and a second time is determined representing when the packet was received. If the difference between the second time and the first time is greater than a predetermined amount, the packet is considered to be stale and is rejected, thereby deterring replay. Because each node maintains its own clock and time, to keep the clocks relatively synchronized, if a time associated with a timestamp of a received packet is later than a certain amount with respect to the time at the receiver, the receiver's clock is set ahead by an amount that expected to synchronize the receiver's and the sender's clocks. However, a receiver never sets its clock back, to deter attacks.

Method for self-synchronizing time between communicating networked systems using timestamps

Multiple aspects of systems and methods for voice control and related features and functionality for various embodiments of media playback devices, networked microphone devices, microphone-equipped media playback devices, and speaker-equipped networked microphone devices are disclosed and described herein, including but not limited to designating and managing default networked devices, audio response playback, room-corrected voice detection, content mixing, music service selection, metadata exchange between networked playback systems and networked microphone systems, handling loss of pairing between networked devices, actions based on user identification, and other voice control of networked devices.

Voice control of a media playback system

In an audio speaker network, setup of speaker location, sound track or channel assignation, and speaker parameters is facilitated by an application detecting speaker locations and prompting a user to input rough room boundaries and a desired listener location in the room. Based on this, optimum speaker locations/frequency assignations/speaker parameters may be determined and output.

Distributed wireless speaker system with automatic configuration determination when new speakers are added

In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) is activated or deactivated as appropriate such that audio play follows a user as the user moves around the home.

Networked speaker system with follow me

Techniques for loudspeaker localization are provided. Sound is received from a loudspeaker at a plurality of microphone locations. A plurality of audio signals is generated based on the sound received at the plurality of microphone locations. Location information is generated that indicates a loudspeaker location for the loudspeaker based on the plurality of audio signals. Whether the generated location information matches a predetermined desired loudspeaker location for the loudspeaker is determined. A corrective action with regard to the loudspeaker is enabled to be performed if the generated location information is determined to not match the predetermined desired loudspeaker location for the loudspeaker.

Loudspeaker localization techniques

Systems and methods are described that automatically adjust a value of a parameter relating to the delivery of media content, such as audio content or image content, based on both environmental conditions and on automatically-learned user preference data. For example, a first embodiment adjusts a volume setting used to control the delivery of an audio signal based both on environmental noise conditions and upon automatically-learned user preference information, wherein the user preference information is derived by monitoring user-implemented adjustments to the volume setting after application of an automatic adjustment thereto. As another example, a second embodiment adjusts a brightness setting used to control the brightness of a display used for rendering images based both on an ambient light level and upon automatically-learned user preference information, wherein the user preference information is derived by monitoring user-implemented adjustments to the brightness setting after application of an automatic adjustment thereto.

Adjustment of media delivery parameters based on automatically-learned user preferences

A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Single microphone wind noise suppression

A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs.

Loudness enhancement system and method

A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Wilfrid LeBlanc

Papers

Loudspeaker localization techniques

Adjustment of media delivery parameters based on automatically-learned user preferences

Single microphone wind noise suppression

Loudness enhancement system and method

Spectral shaping for speech intelligibility enhancement