2919 results about "Audio signal flow" patented technology

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.

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 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.

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.

A technique for enhancing audio signals generated from compressed digital audio files is described. The technique uses a Bass Maximizer module, a Harmonic Exciter module and a Quasi Stereo module. The Bass Exciter module enhances the intensity, depth and punch of the bass audio content by creating harmonic sequences from low frequency components contained in the original input signal. The Harmonic Exciter module adds to the treble audio content of the original input signal by generating harmonic series from the high frequency components contained in the input signal. The Quasi Stereo Module creates a stereo image of the enhanced input signal by adding and subtracting delayed and filtered versions of the enhanced input signal with itself to create left and right channeled stereo-like outputs. The technique provides a useful tool to regenerate from an audio signal more pleasant and joyful sounds.

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.

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.

System and method for visual and audible communication between a central operator and N mobile communicators (N≧2), including an operator transceiver and interface, configured to receive and display, for the operator, visually perceptible and audibly perceptible signals from each of the mobile communicators. The interface (1) presents an audible signal from each communicator as if the audible signal is received from a different location relative to the operator and (2) allows the operator to select, to assign priority to, and to display, the visual signals and the audible signals received from a specified communicator. Each communicator has an associated signal transmitter that is configured to transmit at least one of the visual signal and the audio signal associated with the communicator, where at least one of the signal transmitters includes at least one sensor that senses and transmits a sensor value representing a selected environmental or physiological parameter associated with the communicator.

A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

A method and an apparatus for reproducing a sound signal are provided. The method includes generating an output sound signal to be transmitted to speakers by transmitting a first input sound signal through a filter; acquiring magnitude information of the output sound signal; determining frequency response parameters related to frequency responses of the filter based on the magnitude information; and adaptively adjusting coefficients of the filter based on the determined frequency response parameters.

A method of transferring a real-time audio signal transmission, including: registering voice patterns (or other characteristics) of on more users to be used to identify the voices of the users, accepting an audio signal as it is created as a sequence of segments, analyzing each segment of the accepted audio signal to determine if it contains voice activity (314), determining a probability level that the voice activity of the segment is of a registered user (320&322); and selectively transferring the contents, of a segment responsive to the determined probability level (324).

A system for conducting a virtual audio-visual conference between two or more users comprising two or more client stations each acting as a signal source and destination for each respective user, having a user interface for audio-visual input and output including audio signal reception and generation means for receiving and generating audio signals, one or more servers, and a network coupling the client stations and the servers, wherein each user is represented as a corresponding movable visual symbol displayed on the user interfaces of all coupled client stations and the audio signal of all users is generated at each client station with an attenuation according to the spatial position of the respective symbols on the user interfaces and according to the direction in which each movable visual symbol of each signal source is oriented on the user interface.

A headset having game, chat and microphone audio signals is provided with a programmable signal processor for individually modifying the audio signals and a memory configured to store a plurality of user-selectable signal-processing parameter settings that determine the manner in which the audio signals will be altered by the signal processor. The parameter settings collectively form a preset, and one or more user-operable controls can select and activate a preset from the plurality of presets stored in memory. The parameters stored in the selected preset can be loaded into the signal processor such that the sound characteristics of the audio paths are modified in accordance with the parameter settings in the selected preset.

Scalable coding of audio into a core layer in response to a desired noise spectrum established according to psychoacoustic principles supports coding augmentation data into augmentation layers in response to various criteria including offset of such desired noise spectrum. Compatible decoding provides a plurality of decoded resolutions from a single signal. Coding is preferably performed on subband signals generated according to spectral transform, quadrature mirror filtering, or other conventional processing of audio input. A scalable data structure for audio transmission includes core and augmentation layers, the former for carrying a first coding of an audio signal that places post decode noise beneath a desired noise spectrum, the later for carrying offset data regarding the desired noise spectrum and data about coding of the audio signal that places post decode noise beneath the desired noise spectrum shifted by the offset data.

Provided herein is an electronic device and method of voice recognition, the method including analyzing an audio signal of a first frame when the audio signal is input and extracting a first feature value; determining a similarity between the first feature value extracted from the audio signal of the first frame and a first feature value extracted from an audio signal of a previous frame; analyzing the audio signal of the first frame and extracting a second feature value when the similarity is below a predetermined threshold value; and comparing the extracted first feature value and the second feature value and at least one feature value corresponding to a pre-defined voice signal and determining whether or not the audio signal of the first frame is a voice signal, and thus the electronic device may detect only a voice section from the audio signal while improving the processing speed.

A device comprising a voice activity detector for detecting voice activity in a speech signal using digital data formed on the basis of samples of an audio signal. The voice activity detector comprises a first element adapted to examine whether the signal has a highpass nature. The voice activity detector also comprises a second element adapted to examine the frequency spectrum of the signal. The voice activity detector is adapted to provide an indication of speech when the first element has determined that the signal has a highpass nature or the second element has determined that the signal does not have a flat frequency response.

A noise reduction circuit for reducing the effects of low frequency noise such as wind noise in communications applications is described. In one embodiment, the noise reduction circuit features a high pass filter formed by exploiting the existing off-chip AC coupling capacitances in making the connection to the source of audio signals. The filter may be adaptive to environmental low frequency noise level through programming the shunt resistances. A low-noise wide dynamic range programmable gain amplifier is also described. Adaptive equalization of the audio signal is also described through the utilization of programmable front-end resistors and a back-end audio equalizer.

Devices and methods of detecting a predetermined audio signal in audio signals are provided. A device includes a processor coupled to a clock signal generator, a power controller and an audio detector. The power controller controls a clock rate provided to the processor by the clock signal generator, to control the device to operate in a low power mode having a relatively low power consumption or in a normal power mode having a relatively high power consumption. The audio detector receives audio signals and detects, in the low power mode, probable presence of a predetermined audio signal in the audio signals. The power controller controls the device to switch from the low power mode to the normal power mode responsive to the detected presence of the predetermined audio signal by the audio detector.

A signal transmission apparatus is connected to a network as one node among a plurality of nodes involved in the network which is provided with an audio signal transmission period for transmitting a plurality of channels of audio signals each transmission cycle and a control data transmission period for transmitting control data of the plurality of the nodes each control cycle by using an idle time period other than the audio signal transmission period. In the signal transmission apparatus, a storage section stores configuration information of the one node. A transmitting section transmits the control data including an error checking code of the configuration information. A receiving section receives request data from another node, the request data requesting the one node for transmission of an information block of the configuration information. A control section controls the transmitting section to transmit the information block in response to the request data.

A method for time aligning audio signal, wherein one signal has been derived from the other or both have been derived from another signal, comprises deriving reduced-information characterizations of the audio signals, auditory scene analysis. The time offset of one characterization with respect to the other characterization is calculated and the temporal relationship of the audio signals with respect to each other is modified in response to the time offset such that the audio signals are coicident with each other. These principles may also be applied to a method for time aligning a video signal and an audio signal that will be subjected to differential time offsets.