9944 results about "Audio signal" patented technology

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 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.

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.

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 method and apparatus for intelligently routing and managing audio signals within an electronic device is disclosed. The routing is responsive to a set of logical and physical policies which are stored in data tables which can be updated as needed.

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.

Methods and apparatus for executing a lighting program to control a plurality of light emitting diodes (LEDs) in response to at least one characteristic of an audio input. In one embodiment, the audio input is digitally processed to determine the at least one characteristic. In other embodiments, control signals for the LEDs are generated in response to a timer and/or input from a user interface, as well as in response to the at least one characteristic of the audio input. In another embodiment, the control signals for the LEDs are generated by a same computer that processes the audio input to transmit signals to speakers to audibly play the audio input. In a further embodiment, a GUI is provided to assist in authoring the lighting program. In another embodiment, the audio signal is processed before being played back. In a further embodiment, the lighting program anticipates changes in the audio input.

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.

A weld recording system mounted in or on a welding helmet is provided. The weld recording system includes a camera assembly unit, a power supply unit, a processor, and removable memory. The weld recording system interfaces with lens control circuitry, an optical sensor, a welding power supply, and a helmet position sensor. Logic is provided for the triggering and recording of video and audio signals, which are eventually stored in a file for future reference. Transmission of signals from one or more welders to a monitoring station for eventual display is presented. An image processing algorithm is provided to combine multiple images with varied exposure times into a visual image of the weld and its surroundings.

A method and apparatus for intelligently routing and managing audio signals within an electronic device is disclosed. The routing is responsive to a set of logical and physical policies which are stored in data tables which can be updated as needed.

Audio coding processes like quantization can cause spectral components of an encoded audio signal to be set to zero, creating spectral holes in the signal. These spectral holes can degrade the perceived quality of audio signals that are reproduced by audio coding systems. An improved decoder avoids or reduces the degradation by filling the spectral holes with synthesized spectral components. An improved encoder may also be used to realize further improvements in the decoder.

In accordance with the principles of the present invention, an audio-conference bridging system and method are provided. The present invention discards the traditional notion of a single mixing function for a conference. Instead, the novel, flexible design of the present invention provides a separate mixing function for each participant in the conference. This new architecture is described generally herein as “EdgePoint mixing.” EdgePoint mixing overcomes limitations of traditional conferencing systems by providing each participant control over his/her conference experience. EdgePoint mixing also allows, when desired, the simulation of a “real-life” conference by permitting each participant to receive a distinctly mixed audio signal from the conference depending on the speaker's “position” within a virtual conference world. The present invention also preferably accommodates participants of different qualities of service. Each participant, thus, is able to enjoy the highest-level conferencing experience that his/her own connection and equipment will permit.

Video and audio signals are streamed to remote viewers that are connected to a communication network. A host server receives an originating video and audio signal that may arrive from a single source or from a plurality of independent sources. The host server provides any combination of the originating video and audio signals to viewers connected to a communication network. A viewer requests the host server provide a combination of video and audio signals from the host server. The host server transmits an instruction set to be executed by the viewer. The instruction set causes the viewer to transmit parameters to the host user, including parameters relating to the processing capabilities of the viewer. The host server then transmits multimedia data to the viewer according to the received parameters. A plurality of viewers may be simultaneously connected to the host server. Each of the plurality of viewers may configure the received video and audio signals independent of any other viewer and may generate alerts based on the video and audio content.

Music videos are automatically produced from source audio and video signals. The music video contains edited portions of the video signal synchronized with the audio signal. An embodiment detects transition points in the audio signal and the video signal. The transition points are used to align in time the video and audio signals. The video signal is edited according to its alignment with the audio signal. The resulting edited video signal is merged with the audio signal to form a music video.

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. 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 processing algorithms for processing sound signals received from the hearing aid 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 sources and selective processing 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 processing algorithms for application to selected incoming audio channels and a communication port for the user to connect to a PC or other device to download preferred processing algorithms. The communication port provides the user with the ability to retrieve desirable processing algorithms from a database of available algorithms and download those algorithms directly into the device for use.

Apparatus and methods for including a code having at least one code frequency component in an audio signal are provided. The abilities of various frequency components in the audio signal to mask the code frequency component to human hearing are evaluated and based on these evaluations an amplitude is assigned to the code frequency component. Methods and apparatus for detecting a code in an encoded audio signal are also provided. A code frequency component in the encoded audio signal is detected based on an expected code amplitude or on a noise amplitude within a range of audio frequencies including the frequency of the code component.

Methods and systems for determining and utilizing spatial correlative information relating to two or more devices to determine device positioning are described. Using audio signals emitted from stereo speakers, for example, associated with a first device and a microphone associated with the second device, the distance and angle between the two devices and as their relative positions can be determined. No other sensors or specialized accessories are needed on either device to calculate the distance and angles. The devices need only be loaded with the appropriate software which, when executed, is able to carry out steps of the present invention. The usefulness of one or both of the devices may be enhanced by knowing the distance and angle data between the devices. For example, one device may be a TV having stereo speakers and the other device may be a handheld device, such as a smartphone, having a microphone.

A system and method are provided for smoothing the in-room acoustic magnitude response of an audio reproduction system. An in-room acoustic magnitude response analysis is performed to determine a room resonance induced peak associated with an audio signal. A replica of the audio signal is filtered at the room resonance induced peak. The filtered replica signal is added with the audio signal. Through this, smoothing of the room resonance induced peak may be achieved, such that a subjective impression of transient response and dynamics of the audio signal are preserved.

An audio system to calibrate an audio signal based on a wirelessly received signal, and a signal calibration method are provided. The audio system includes a sound output unit to output a sound corresponding to a received audio signal. A transceiver is connected to an external device to enable wireless communication between a main unit and the external device. The external device converts the sound output from the sound output unit into an electric signal to generate a calibration audio signal. The main unit performs calibration on an audio signal to be played back through the sound output unit using the calibration audio signal.

A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.