Abstract: The invention relates to a method and an apparatus for processing an audio signal, wherein the method comprises the steps of: filtering an audio signal into at least two frequency band signals, generating for each frequency band signal a plurality of sub-band signals, wherein for at least one frequency band signal the plurality of sub-band signals are generated using a time to frequency domain transform and for the at least one other frequency band the plurality of sub-band signals for the other frequency band are generated using a sub-band filterbank, and the apparatus comprises at least one processor and at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processor, cause the apparatus to perform the method.

Abstract: Methods and apparatus for detecting a repetitive pattern in a sequence of audio frames are described. Similarity values of a first similarity matrix with first resolution for the sequence are calculated. An adaptive threshold is estimated from the similarity values for classifying the similarity values into repetition or non-repetition. For each of one or more offsets of a second similarity matrix with second resolution higher that the first resolution, similarity values of the second similarity matrix corresponding to the offset are calculated. Then the calculated similarity values are binarized with the adaptive threshold to obtain binarized data. Finally, the repetitive pattern is detected from the binarized data. The requirement on memory may be reduced because less data are stored in detecting the repetitive pattern.

Abstract: Various methods are described for generating an audio summary representing a location on a place exploration service. One example method may comprise receiving at least one audio file. The method may further comprise dividing the at least one audio file into one or more audio segments. Additionally, the method may comprise determining a representative audio segment for each of the one or more audio segments. The method may further comprise generating an audio summary of the at least one audio file by combining one or more of the representative audio segments of the one or more audio segments. Similar and related methods, apparatuses, and computer program products are also provided.

Abstract: Loud sounds with fast rise times, like gunfire and explosions, can cause noise-induced hearing loss (NIHL). Unfortunately, current models do not adequately explain how impulsive sounds cause NIHL, which makes it difficult to predict and prevent NIHL on battlefields and other hostile or rugged environments. Fortunately, the impulsive sounds experienced by soldiers and others working in rugged environments can be recorded using a compact, portable system that acquires, digitizes, and stores high-bandwidth audio data. An example of this system can be mounted on a helmet or other article and used to record hours of audio data at a bandwidth of 20 kHz or higher, which is broad enough to capture sounds with rise times less than 50 ms. An analog-to-digital converter (ADC) digitizes these broadband audio signals at rate of 40 kHz or higher to preserve the impulse information. A processor transfers the digitized samples from a buffer to a memory card for later retrieval using an interrupt-driven processing technique.

Abstract: Systems and methods are provided herein relating to audio matching. Interest points that are onsets are generally very efficient in audio matching in that they are robust to multiple types of distortion. Prominent onsets can be detected within an audio signal excerpt as interest points and combined as a function of a set of interest points to form a descriptor. Descriptors associated with an audio signal excerpt that contain a set of prominent onsets as interest points can be used in matching the audio signal excerpt to an audio reference. The benefits in generating and using prominent onsets within descriptors improve the accuracy of an audio matching system.

Abstract: A method and system for modifying an audio signal, the method comprising: receiving the audio signal at signal processing means; analysing the received audio signal to identify characteristic signal components in the audio signal; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively applying an adjusted gain to the audio signal based on the determination that the input signal is input to the signal processing means, wherein the adjusted gain is generated in dependence upon the signal strength of the identified signal components.

Abstract: The invention concerns an audio signal processing unit including a receiving unit adapted to receive at least a first audio signal and a pilot tone signal, a first audio signal amplifier adapted to obtain at least a second audio signal from an audio signal source by way of an audio signal input, and a pilot tone signal unit adapted to obtain the received pilot tone signal. The pilot tone signal unit is adapted to output a control signal in a first state if the pilot tone signal is received, and to output the control signal in a second state if the pilot tone signal is not received. The first audio signal amplifier is adapted to process the at least second audio signal in dependence on the state of the control signal.

Abstract: A system to remaster an audio recording by processing selected frequency bands and reducing the volume of each frequency band at selected points during processing.

Abstract: The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Abstract: An audio signal distortion detection device for simplifying the detection of harmonic distortion to aid in setting an amplifier's gain control to match an audio source unit's output voltage. The audio signal distortion detection device generally includes a housing having a plurality of status indicators thereon for indicating the status of an audio signal being fed into the housing's audio input. By utilizing a scaling circuit and processing unit, the present invention may be configured to detect four audio frequencies and compare DC levels of the frequencies to each other to determine if harmonic distortion is present. A distortion illuminator will illuminate when such distortion is detected and thus significantly reduce the effort required to setting an amplifier's gain control.

Abstract: A cable for transmitting a transmissible element, such as power or data, from a source to a device, the cable includes at least a first plug configured to mate with at least one of the source and the device, and an audio player for playing back audio recording.

Abstract: A method and apparatus to encode audio with spatial information in a manner that does not depend on the exhibition setup, and to decode and play out optimally for any given exhibition setup, maximizing the sweet-spot area, and including setups with loudspeakers at different heights, and headphones. The part of the audio that requires very precise localization is encoded into a set of mono tracks with associated directional parameters, whereas the remaining audio is encoded into a set of Ambisonics tracks of a chosen order and mixture. Upon specification of a given exhibition system, the exhibition-independent format is decoded adapting to the specified system, by using different decoding methods for each assigned group.

Abstract: A system for assisting an end user to develop a musical composition comprising a data storage medium, a plurality of musical assets stored in the data storage medium, a user interface, and an audio converter module. Each of the plurality of musical assets are associated with a functional attribute and a visual attribute, where the visual attribute provides a visual suggestion to an end user regarding the functional attribute of the musical asset. The user interface is coupled to an audio interface, a display driver and a user input handler; and is capable of rendering a graphical display that enables the end user to select between one or more visual attributes. The audio converter module captures audio recorded by the end-user through the audio interface, processes the audio; and produces a multi-track musical compilation comprised of a plurality of audio tracks mixed together in a manner where the multiple tracks are substantially musically harmonious with one another.

Abstract: Sequential digital audio signals are received to calculate a difference between each currently sampled digital audio signal and another digital audio signal sampled at one sampling period before each currently sampled digital audio signal. Differences for the sequential digital audio signals are stored. The number of digital audio signals consecutively clipped is counted in the received sequential digital audio signals. A specific difference is retrieved, from the stored differences, for a digital audio signal sampled at a specific number of sampling periods before each clipped digital audio signal. The specific number of sampling periods is determined based on the counted number of digital audio signals consecutively clipped. Each clipped digital audio signal is corrected based on the specific difference.

Abstract: Apparatus, systems and methods provide incentive-based usage of an audio recognition system. In an aspect, a system is provided that includes a query component configured to receive an audio sample from a device and a recognition component configured to determine an identification of the audio sample. The system further includes a reward component configured to identify a reward associated with the identification of the audio sample, wherein the query component is further configured to provide a query result to the device, the query result comprising the identification of the audio sample and the reward associated therewith.

Abstract: An audio system has a connector with a signal pin that interfaces the audio system with an external device. An audio integrated circuit has a digital to analog converter (DAC) and an audio power amplifier having an input coupled to an output of the DAC and an output coupled to an output pin of the audio integrated circuit. Data storage contains stored instructions that program a processor to select an audio output mode of operation namely line-out mode or headphone mode. A configurable output conditioning circuit is coupled between the output pin of the audio integrated circuit and the signal pin of the connector. The conditioning circuit changes the output impedance that is presented at the signal pin, under control of the programmed processor when selecting the audio output mode of operation. Other embodiments are also described and claimed.

Abstract: The present invention relates to an audio signal controller adapted to receiving first and second digital audio signals and estimating a signal feature of the first or second digital audio signal. The estimated signal feature is compared with a predetermined feature criterion and the audio signal controller switches from conveying the first digital audio signal to conveying the second digital audio signal to a controller output, or vice versa, at a zero-crossing of the first digital audio signal or the second digital audio signal based on the comparison between the estimated signal feature and the predetermined feature criterion.

Abstract: Method and devices for testing a headphone with increased sensation are provided. The headphone can filter and amplify low frequency audio signals, which are then sent to a haptic device in the headphone. The haptic device can cause bass sensations at the top of the skull and at both ear cups. The testing system can evaluate the haptic and acoustic sensations produced by the headphone to evaluate if they have been properly assembled and calibrate the headphones if necessary.

Abstract: In an active muffler having improved response characteristics, a speaker section includes a diaphragm adapted to generate sound, a voice coil for driving the diaphragm, and a distance sensor to detect the movement of the diaphragm. A light generated by the LED is reflected by the diaphragm, the reflected light is detected by a phototransistor to thereby measure the distance to the diaphragm, so that the movement of the diaphragm is detected. Noise is detected by a microphone, and a signal having opposite phase to that of the noise is generated by an opposite-phase generating section. The difference between the opposite-phase signal and the signal of the distance to the speaker from the distance sensor is calculated and inputted to a PID control section. Such a difference indicates the delay of the speaker movement. Feedback control is performed in a direction in which the difference is canceled out.

Abstract: Method and devices for processing audio signals based on sound profiles are provided. A reproduction device can request a sound profile based on user information, device information, media metadata or a combination. The sound profiles can be customized and shared across multiple devices. User interfaces allow for the input of information that allows the reproduction device or a server in the cloud to select, modify, store, and analyze sound profiles. Deeper analysis allows for the improvement of sound profiles for individuals and groups. Intensity scoring of a music library can also be conducted.