10838results about "Gain control" patented technology

The present invention is directed to a method and system for automatic calibration of an acoustic system. The acoustic system may include a source A / V device, calibration computing device, and multiple rendering devices. The calibration system may include a calibration component attached to each rendering device and a source calibration module. The calibration component on each rendering device includes a microphone. The source calibration module includes distance and optional angle calculation tools for automatically determining a distance between the rendering device and a specified reference point upon return of the test signal from the calibration component.

The various methods and devices described herein relate to devices which, in at least certain embodiments, may include one or more sensors for providing data relating to user activity and at least one processor for causing the device to respond based on the user activity which was determined, at least in part, through the sensors. The response by the device may include a change of state of the device, and the response may be automatically performed after the user activity is determined.

Various dynamic audio ducking techniques are provided that may be applied where multiple audio streams, such as a primary audio stream and a secondary audio stream, are being played back simultaneously. For example, a secondary audio stream may include a voice announcement of one or more pieces of information pertaining to the primary audio stream, such as the name of the track or the name of the artist. In one embodiment, the primary audio data and the voice feedback data are initially analyzed to determine a loudness value. Based on their respective loudness values, the primary audio stream may be ducked during the period of simultaneous playback such that a relative loudness difference is generally maintained with respect to the loudness of the primary and secondary audio streams. Accordingly, the amount of ducking applied may be customized for each piece of audio data depending on its loudness characteristics.

An interface which allows a surgeon to operate multiple surgical devices from a single input device. The input device may be a foot pedal that provides output signals to actuate a number of different surgical devices. The surgical devices may include a robotic arm, a laser, an electrocautery device, or an operating table. The interface has an input channel that is coupled to the input device and a plurality of output channels that are coupled to the surgical devices. The interface also has a select input channel which can receive input commands to switch the input channel to one of the output channels. The select channel may be coupled to a speech interface that allows the surgeon to select one of the surgical devices with a voice command. The surgeon can operate any device by providing an input command which switches the input channel to the desired output channel.

An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

One or more acoustic transducers of a device may be adjusted based on automatic detection of device proximity to the user. In a mobile telephone, when the user is using the receiver and holding the telephone against his / her ear, if the telephone detects that the user has moved the telephone further from his / her ear, the telephone will raise the receiver volume. Similarly, if the user is using the speaker, the telephone will adjust the speaker volume as user distance from the telephone changes. In another embodiment the telephone may fade between the receiver and the speaker. Volume is not the only acoustic property that could be adjusted according to user proximity. Frequency response is another property that could be adjusted, such as using appropriate electronic filtering, or by turning on another transducer that is not otherwise being used.

An audio user interface that provides audio prompts that help a user interact with a user interface of an electronic device is disclosed. The audio prompts can provide audio indicators that allow a user to focus his or her visual attention upon other tasks such as driving an automobile, exercising, or crossing a street, yet still enable the user to interact with the user interface. An intelligent path can provide access to different types of audio prompts from a variety of different sources. The different types of audio prompts may be presented based on availability of a particular type of audio prompt. As examples, the audio prompts may include pre-recorded voice audio, such as celebrity voices or cartoon characters, obtained from a dedicate voice server. Absent availability of pre-recorded or synthesized audio data, non-voice audio prompts may be provided.

A method (200) for improving quality of output audio (126). The method can include detecting an output acoustic signal (128) and generating a receive audio signal (134) based, at least in part, on the detected output acoustic signal. A frequency domain representation (140) of the receive audio signal can be compared to a frequency domain representation (138) of a source audio signal (124) from which the output acoustic signal is generated. At least one distortion signal (142) in the receive audio signal can be identified, and the source audio signal can be selectively equalized to reduce an amplitude of the source audio signal at a frequency that correlates to the distortion signal.

Improved techniques for presenting sound effects at a portable media device are disclosed. The sound effects can be output as audio sounds to an internal speaker, an external speaker, or both. In addition, the audio sounds for the sound effects can be output together with other audio sounds pertaining to media assets (e.g., audio tracks being played). In one embodiment, the sound effects can serve to provide auditory feedback to a user of the portable media device. A user interface can facilitate a user's selection of sound effect usages, types or characteristics.

One embodiment of the present invention provides a system that uses a sound sensor to adjust the audio output of a device. During operation, the system uses the sound sensor to determine an ambient sound level for the environment in the proximity of the device. The system then adjusts a volume setting for the device adaptively based on the determined ambient sound level. Adaptively adjusting the volume setting allows the device to adapt to its audio environment and ameliorates potentially-disruptive audio outputs.

A method of compensating for spatial audio frequency characteristics that varies in accordance with a mounting condition of a down firing speaker of an audio / video (AV) apparatus includes calculating a listening distance between the AV apparatus and a listener, calculating a distance between a speaker mounted on the AV apparatus and a neighboring reflective surface, setting a spatial frequency compensation filter value and a speaker frequency characteristic compensation filter value based on the calculated distances, and compensating for frequency characteristics of an audio signal by combining the spatial frequency compensation filter value and the speaker frequency characteristic compensation filter value.

An amplification architecture or system is provided having a multiple amplifier system that switches modes of operation between operation in a component mode and a composite mode based on a characteristic of an input signal relative to a threshold level. In the component mode, the components of the input signal are employed to different terminals of the multiple amplifier system that provide a reconstructed amplified representation of the input signal. In the composite mode, the input signal is amplified to provide an amplified representation of the input signal.

A network-enabled audio device that provides a display device that allows the user to select playlists of music much like a jukebox is disclosed. The user can compose playlists from disk files, CD's, Internet streaming audio broadcasts, online music sites, and other audio sources. The user can also select a desired Web broadcast from a list of available Web broadcasts. In addition, the user can play standard audio CD's and MP3 encoded CD's and have access to local AM / FM stations. Further, the software, the user controls, and the display in the network-enabled audio device are operably configured and connected such that the user can listen to playlists that include CD's and other audio sources just as the user would choose a playlist in a jukebox. The user accesses a server site via a PC and the Internet. From the server site, the user obtains a list of the devices in his or her Internet Personal Audio Network (IPAN) and what songs are on those devices. The IPAN includes an IPAN server, an IPAN client, and IPAN software stored on the network-enabled audio device. Thus, the network-enabled audio device provides people who are or are not comfortable with computers a way of taking music from various sources and putting it into one place for listening pleasure. In one embodiment, the Personal Computer (PC) is used to compose the playlists, but the user is able to listen to playlists and other audio sources without using the PC.

Exemplary systems and methods include a distribution device that maintains a dock rate and distributes a series of tasks to a group of execution devices. Each task has a plurality of samples per frame associated with a time stamp indicating when the task is to be executed. The execution devices execute the series of tasks at the times indicated and adjust the number of samples per frame in relation to the dock rate maintained by the distribution device.

Some embodiments of the present invention provide a method of adjusting an output of a media player comprising capturing an ambient audio signal; processing the ambient audio signal to determine whether one or more characteristic forms are present within the ambient audio signal; and reducing an output of a media player from a first volume to a second volume if the one or more characteristic forms are present within the ambient audio signal. The characteristic forms may be, for example, a name or personal identifier of a user of the media player, the voice of a user of the media player, or an alarm or siren.