25501 results about "Audio signal" patented technology

A comprehensive, wireless multimedia surveillance and monitoring system provides a combination of megapixel digital camera capability with full motion video surveillance with a network, including network components and appliances such as wiring, workstations, and servers with the option of geographical distribution with various wide area carriers. The full service, multi-media surveillance system is capable of a wide range of monitoring techniques utilizing digital network architecture and is adapted for transmitting event data, video and / or image monitoring information, audio signals and other sensor and detector data over significant distances using digital data transmission over a LAN, wireless LAN, Intranet or Internet for automatic assessment and response including dispatch of response personnel. Both wired and wireless appliance and sensor systems may be employed. GPS dispatching is used to locate and alert personnel as well as to indicate the location of an event. Automatic mapping and dispatch permits rapid response. The wireless LAN connectivity permits local distribution of audio, video and image data over a relatively high bandwidth without requirement of a license and without relying on a common carrier and the fees associated therewith. The surveillance system may be interfaced with a WAN (wide area Network) or the Internet for providing a worldwide, low cost surveillance system with virtually unlimited geographic application. Centralized monitoring stations have access to all of the surveillance data from various remote locations via the Internet or the WAN. A server provides a centralized location for data collection, alarm detection and processing, access control, dispatch processing, logging functions and other specialized functions. The server may be inserted virtually anywhere in the Intranet / Internet network. The topology of the network will be established by the geographic situation of the installation. Appropriate firewalls may be set up as desired. The server based system permits a security provider to have access to the appliance and sensor and surveillance data or to configure or reconfigure the system for any station on the network.

A method for recognizing an audio sample locates an audio file that most closely matches the audio sample from a database indexing a large set of original recordings. Each indexed audio file is represented in the database index by a set of landmark timepoints and associated fingerprints. Landmarks occur at reproducible locations within the file, while fingerprints represent features of the signal at or near the landmark timepoints. To perform recognition, landmarks and fingerprints are computed for the unknown sample and used to retrieve matching fingerprints from the database. For each file containing matching fingerprints, the landmarks are compared with landmarks of the sample at which the same fingerprints were computed. If a large number of corresponding landmarks are linearly related, i.e., if equivalent fingerprints of the sample and retrieved file have the same time evolution, then the file is identified with the sample. The method can be used for any type of sound or music, and is particularly effective for audio signals subject to linear and nonlinear distortion such as background noise, compression artifacts, or transmission dropouts. The sample can be identified in a time proportional to the logarithm of the number of entries in the database; given sufficient computational power, recognition can be performed in nearly real time as the sound is being sampled.

A power delivery method and system for powering a headset. A power signal is combined with an audio signal to form a composite signal that is communicated over a shared channel to the headset. The power signal is generated by modulating a carrier signal with a modulation signal. The modulation signal is derived from the amplitude of the audio signal so that the peak levels of the composite signal do not exceed the maximum allowable output of an audio I / O circuit driving the headset.

A method and apparatus for providing streaming content over telephones is described. The creation of a voice portal is supported by the invention. Embodiments of the invention use allows users to place a telephone call to access the voice portal. The user can access many different types of content. This content can include text based content which is read to the user by a text to speech system (e.g., news reports, stock prices, text content of Internet sites), audio content which can be played to the user (e.g., voicemail messages, music), and streaming audio content (e.g., Internet broadcast radio shows, streaming news reports, and streaming live broadcasts). This content can be accessed from many different places. For example, the content can be retrieved from a news feed, a local streaming content server, an audio repository, and / or an Internet based streaming content server. The streaming content allows the user to access live web broadcasts even though the user may not have access to a computer. The system allows audio signals from multiple sources to be combined to enhance the user experience. Additionally, the user can control many aspects of the delivery of the streaming content. Personalization features of the voice portal allow for an improved user experience during streaming content delivery.

Position information of equipment at an event, such as a ball, one or more players, or other items in a game or sporting event, is used in selecting camera, camera shot type, camera angle, audio signals, and / or other output data for providing a multimedia presentation of the event to a viewer. The position information is used to determine the desired viewer perspective. A network of feedback robotically controlled Pan-Tilt-Zoom (PTZ), manually controlled cameras and stationary cameras work together with interpolation techniques to create a 2D video signal. The position information may also be used to access gaming rules and assist officiating of the event. The position information may be obtained through a transceiver(s), accelerometer(s), transponder(s), and / or RADAR detectable element(s) fitted into the ball, apparel or equipment of players, the players themselves, or other playing equipment associated with the game or sporting event. Other positioning methods that can be used include infrared video-based tracking systems, SONAR positioning system(s), LIDAR positioning systems, and digital signal processing (DSP) image processing techniques such as triangulation.

A music recommendation system receives a user selection of desired music, retrieves analysis data associated with the selected music, and generates a playlist of songs based on the analysis data. The analysis data is generated based on a processing of one or more audio signals associated with the selected music. The analysis data may downloaded from a central server. If the analysis data is not available from the central server, it is generated locally at a user end, and uploaded to the central server. A plurality of user-selectable shuffling mechanisms are provided to allow the order of the songs to be shuffled according to the selected shuffling mechanism. The end user device may also receive recommendation of new music from different providers based on the analysis data of music for which the recommendation is to be based.

A video-audio surveillance system for a motor vehicle is suitable for observing blind spots as well as for providing security for the vehicle and its contents. The system includes audio-video transmission modules positioned at selected locations on the exterior and interior of the vehicle. An operator or observer either in the vehicle or otherwise monitors the video and audio signals from the modules through a monitoring unit.

A method and system is described which allows users to identify (pre-recorded) sounds such as music, radio broadcast, commercials, and other audio signals in almost any environment. The audio signal (or sound) must be a recording represented in a database of recordings. The service can quickly identify the signal from just a few seconds of excerption, while tolerating high noise and distortion. Once the signal is identified to the user, the user may perform transactions interactively in real-time or offline using the identification information.

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.

A routing screen is presented on an electronic device by a user interface application in response to receiving a notification that an external audio device is connected to the electronic device. The routing screen displays representations of an internal audio device and the external audio device. In one aspect, the representations are buttons. In another aspect, the representations are entries in a list. If a user selects one of representations, the user interface application causes the audio signals to be routed to the audio device represented by the selection. An application control screen having a set of objects that represent functions for an audio application may also be displayed. One of the objects on the application control screen is modified in response status changes in the external audio device. A user may select this object to access the routing screen when the external audio device is connected.

Interactive digital media platform, methods and apparatus for detecting and dynamically synchronizing to media content (e.g., television (TV) programs or movies) that a viewer is watching while providing related content on a second screen for enhancing the viewer experience. In one embodiment, the primary content is determined by detecting an audio signal of the primary content via the second screen device; the audio signal may then be processed to generate a fingerprint for comparison with a data store of primary content. The primary content can be classified by various categories (e.g., unique program, advertising, repeat airing, theme song . . . ) and the classification used to aid in the identification and / or in selection of the content to be presented on the interactive second screen device. The system allows a substantially real time comparison and recognition of what primary content a viewer is watching on a first screen device and presentation to the user of content that is substantially synchronous to the viewer's location in the primary content. The viewer can actively engaged with the content presented and can share the content with others via social networking and the like.

Apparatus is described for digitally encoding an input audio signal for storage or transmission. A distinguishing parameter is measure from the input signal. It is determined from the measured distinguishing parameter whether the input signal contains an audio signal of a first type or a second type. First and second coders are provided for digitally encoding the input signal using first and second coding methods respectively and a switching arrangement directs, at any particular time, the generation of an output signal by encoding the input signal using either the first or second coders according to whether the input signal contains an audio signal of the first type or the second type at that time. A method for adaptively switching between transform audio coder and CELP coder, is presented. In a preferred embodiment, the method makes use of the superior performance of CELP coders for speech signal coding, while enjoying the benefits of transform coder for other audio signals. The combined coder is designed to handle both speech and music and achieve an improved quality.

In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information. Since the channel side information only occupy a low number of bits, and since the decoder does not use dematrixing, an efficient and high quality multi-channel extension for stereo players and enhanced multi-channel players is obtained.

Hashes are short summaries or signatures of data files which can be used to identify the file. Hashing multimedia content (audio, video, images) is difficult because the hash of original content and processed (e.g. compressed) content may differ significantly. The disclosed method generates robust hashes for multimedia content, for example, audio clips. The audio clip is divided (12) into successive (preferably overlapping) frames. For each frame, the frequency spectrum is divided (15) into bands. A robust property of each band (e.g. energy) is computed (16) and represented (17) by a respective hash bit. An audio clip is thus represented by a concatenation of binary hash words, one for each frame. To identify a possibly compressed audio signal, a block of hash words derived therefrom is matched by a computer (20) with a large database (21). Such matching strategies are also disclosed. In an advantageous embodiment, the extraction process also provides information (19) as to which of the hash bits are the least reliable. Flipping these bits considerably improves the speed and performance of the matching process.

A speaker includes at least one transducer and at least one memory device. The at least one transducer is adapted to receive an audio signal. The at least one memory device is adapted to store data related to the speaker. A method includes reading data from a memory device of at least one speaker. An audio signal is provided from an audio system.

An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6). Applications in high fidelity audio reproduction, and in car stereo reproduction.

Systems and methods are provided for decoding a message symbol in an audio signal. This message symbol is represented by first and second code symbols displaced in time. Values representing the code signals are accumulated and the accumulated values are examined to detect the message symbol.

A sound produced at the location of a listener is captured by a microphone in each of a plurality of speaker devices. A sever apparatus receives an audio signal of the captured sound from all speaker devices, and calculates a distance difference between the distance of the location of the listener to the speaker device closest to the listener and the distance of the listener to each of the plurality of speaker devices. When one of the speaker devices emits a sound, the server apparatus receives an audio signal of the sound captured by and transmitted from each of the other speaker devices. The server apparatus calculates a speaker-to-speaker distance between the speaker device that has emitted the sound and each of the other speaker devices. The server apparatus calculates a layout configuration of the plurality of speaker devices based on the distance difference and the speaker-to-speaker distance.

A listener can experience the sound of virtual loudspeakers over headphones with a level of realism that is difficult to distinguish from the real loudspeaker experience. Sets of personalized room impulse responses (PRIRs) are acquired for the loudspeaker sound sources over a limited number of listener head positions. The PRIRs are then used to transform an audio signal for the loudspeakers into a virtualized output for the headphones. Basing the transformation on the listener's head position, the system can adjust the transformation so that the virtual loudspeakers appear not to move as the listener moves the head.

A device comprise an a display screen and an audio circuit for generating an audio signal representing spoken words uttered by the user. A processor executes a first application, a second application, and a text mark-up object. The first application may render a depiction of text on the display screen. The text mark-up object may: i) receiving at least a portion of the audio signal representing spoken words uttered by the user; ii) performing speech recognition to generate a text representation of the spoken words uttered by the user; iii) determining a selected text segment, and iv) performing an input function to input the selected text segment to the second application. The selected text segment may be text which corresponds to both a portion of the depiction of text on the display screen and the text representation of the spoken words uttered by the user.

An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

An audio encoder applies an adaptive block-encoding process to segments of audio information to generate frames of encoded information that are aligned with a reference signal conveying the alignment of a sequence of video information frames. The audio information is analyzed to determine various characteristics of the audio signal such as the occurrence and location of a transient, and a control signal is generated that causes the adaptive block-encoding process to encode segments of varying length. A complementary decoder applies an adaptive block-decoding process to recover the segments of audio information from the frames of encoded information. In embodiments that apply time-domain aliasing cancellation (TDAC) transforms, window functions and transforms are applied according to one of a plurality of segment patterns that define window functions and transform parameters for each segment in a sequence of segments. The segments in each frame of a sequence of overlapping frames may be recovered without aliasing artifacts independently from the recovery of segments in other frames. Window functions are adapted to provide preferred frequency-domain responses and time-domain gain profiles.

A body worn communications device for communicating with a head-worn listening device, the communications device being adapted for receiving a multitude of audio signals and for transmitting at least one audio signal selected among the multitude of audio signals to the listening device, the communications device having a number of functional push-buttons for influencing the selection and properties of said audio signals. The communications device has a user interface having a number of functional push-buttons for influencing the state of the user interface, such as the selection (and de-selection) of an audio signal, events and properties related to the audio signal, and wherein the state of the user interface is indicated at the same button where the state can be influenced.

A portable assistive listening system for enhancing sound for hearing impaired individuals includes a fully functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the present invention is directed to the handheld DSP device and a unique method of processing incoming audio signals. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and / or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different sound enhancement algorithms for enhancing sound signals received from the hearing aid and / or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sound sources and selective enhancement of sound. In the context of being user programmable, the digital signal processing device includes a software platform that provides for the ability of the user to select, or plug-in, desired enhancement algorithms for application to selected incoming audio signals. Specifically, the invention focuses on a method of buffering an incoming audio signal, and selectively replaying the buffered audio. The method further includes converting the replayed audio signal to text for display on the handheld DSP device.