Abstract: An exemplary system comprises a device including a memory with an audio injection application installed thereon. The application comprises an equalizer module that analyzes sound characteristics of individual digital audio samples including a discrete signal, a selector module that applies a selection heuristic to select the discrete signal from the individual digital audio samples based on the sound characteristics, and an audio module that supplies to an output an insert signal generated according to the discrete signal selected by the selection heuristic.

Abstract: A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Abstract: Systems and methods are disclosed for providing portable device presence utilizing environmental ambient audio fingerprints. Portable devices provide media exposure data and environmental ambient fingerprints to a processing device, where the environmental ambient signatures provide at least one characteristic of the ambient audio surrounding each portable device. The environmental ambient signatures are then processed to determine if they match. Portable devices associated with matching signatures are identified. Ambient fingerprints may also be used to establish a logical location where media exposure took place. The ambient signatures are alternately combined with monitored data to provide more robust data sets for contextual processing.

Abstract: An adaptive noise canceling (ANC) circuit adaptively generates an anti-noise signal from a reference microphone signal that is injected into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone proximate the speaker provides an error signal. A secondary path estimating adaptive filter estimates the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Tones in the source audio, such as remote ringtones, present in downlink audio during initiation of a telephone call, are detected by a tone detector using accumulated tone persistence and non-silence hangover counting, and adaptation of the secondary path estimating adaptive filter is halted to prevent adapting to the tones. Adaptation of the adaptive filters is then sequenced so any disruption of the secondary path adaptive filter response is removed before allowing the anti-noise generating filter to adapt.

Abstract: An apparatus and method for transmitting a plurality of audio objects using a multichannel encoder and a multichannel decoder are provided. The audio object encoder includes a multichannel encoder determination unit to determine a multichannel encoder to be used for encoding of a plurality of audio objects according to the number of the audio objects, an encoding unit to generate an encoded signal by encoding the plurality of audio objects using the determined multichannel encoder, and a multichannel audio object signal generation unit to generating a multichannel audio object signal, by multiplexing sound image localization information of the plurality of audio objects along with the encoded signal.

Abstract: A communication system reduces power consumption of a radio receiver and/or ensures quality of signal transmission by using various modes of receiver operation. In one embodiment, the communication system is part of a hearing assistance system with a hearing aid including the radio receiver. In one embodiment, a mode of reception is selected during signal transmission based on a detected parameter indicative of quality of the signal transmission. In one embodiment, a hearing aid includes a radio receiver using independent sidebands to receive signals transmitted using each of an upper-sideband and a lower-sideband.

Abstract: There is described discovery of a plurality of audio devices, and for the discovered audio devices are determined relative positions thereof and distances therebetween. The determined relative positions and distances are used to select a constellation of audio devices from the discovered plurality. This constellation is selected for playing or recording of a multi-channel audio file so as to present an audio effect such as a spatial audio effect. Channels for the multi-channel audio file are allocated to different audio devices of the selected constellation, which are controlled to synchronously play back or record their respectively allocated channel or channels of the multi-channel audio file. In a specific embodiment the determined distances are used to automatically select the constellation and include distance between each pair of audio devices of the plurality. Several embodiments are presented for determining the distances and relative positions.

Abstract: A computer-implemented method for reducing a noise signal added to an amplitude modulated (AM) broadcast signal while travelling from a broadcasting antenna to a receiving antenna is provided. The method includes capturing a signal representative of the AM broadcast signal corrupted by the noise signal via the receiving antenna, inverting the captured signal, and determining a carrying frequency of the AM broadcast signal and delaying the inverted waveform by a fraction of a cycle of the carrying frequency. The method further includes generating a difference signal by subtractively combining the captured signal and the delayed inverted signal, generating an estimate noise signal by reducing an amplitude of the generated difference signal using a noise-reduction control multiplier, and minimizing the corrupting noise signal component of the captured signal by subtractively combining the captured signal and the generated estimate noise signal.

Abstract: An apparatus for modifying an audio signal has an envelope shape determiner, a filterbank processor, a signal processor, a combiner and an envelope shaper. The envelope shape determiner determines envelope shape coefficients based on the a frequency domain audio signal representing a time domain input audio signal and the filterbank processor generates a plurality of bandpass signals in a subband domain based on the frequency domain audio signal. Further the signal processor modifies a subband domain bandpass signal of the plurality of subband domain bandpass signals based on a predefined modification target. The combiner combines at least a subset of the plurality of subband domain bandpass signals containing the modified subband domain bandpass signal to obtain a time domain audio signal. Further, the envelope shaper is operative to obtain a shaped audio signal.

Abstract: According to one embodiment, a sound localization apparatus includes a storage unit, a selection unit, and a first operation unit. The storage unit stores a plurality of acoustic transfer characteristics each corresponding to a sound image direction and an emphasis degree of feeling of localization. The selection unit is configured to select a suitable acoustic transfer characteristic from the plurality of acoustic transfer characteristics. The suitable acoustic transfer characteristic is most suitable for the sound image direction indicated by a direction indication information and the emphasis degree indicated by an emphasis degree indication information. The first operation unit is configured to convolute the suitable acoustic transfer characteristic with a first audio signal to obtain a second audio signal.

Abstract: Methods, apparatus, and computer-readable media are described herein related to using self-generated sounds for determining a worn state of a wearable computing device. A wearable computing device can transmit an audio signal. One or more sensors coupled to the wearable computing device may then receive a modified version of the audio signal. A comparison may be made between the modified version of the audio signal and at least one reference signal, where the at least one reference signal is based on the audio signal that is transmitted. Based on an output of the comparison, a determination can be made of whether the wearable computing device is being worn.

Abstract: A reverberator for reverberating an audio signal includes a feedback delay loop processor for delaying at least two different frequency subband signals representing the audio signal by different loop delays to obtain reverberated frequency subband signals.

Abstract: In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) is activated or deactivated as appropriate such that audio play follows a user as the user moves around the home.

Abstract: Methods and apparatus to identify a type of acoustic event Mel Frequency Cepstral Coefficients (MFCCs). In one embodiment, received sound is processed using the MFCCs to locate a peak for comparison with a stored event to determine if the peak corresponds to the acoustic event.

Abstract: An apparatus for modifying an audio signal includes a filterbank processor, a fundamental determiner, an overtone determiner, a signal processor and a combiner. The filterbank processor generates a plurality of bandpass signals based on an audio signal and the fundamental determiner selects a bandpass signal of the plurality of bandpass signals to obtain a fundamental bandpass signal. The overtone determiner identifies a bandpass signal of the plurality of bandpass signals fulfilling an overtone criterion regarding the selected fundamental bandpass signal to obtain an overtone bandpass signal associated to the selected fundamental bandpass signal. The signal processor modifies the selected fundamental bandpass signal based on a predefined modification target. Additionally, the signal processor modifies an identified overtone bandpass signal associated to the selected fundamental bandpass signal depending on the modification of the selected fundamental bandpass signal.

Abstract: An audio unit mountable in a personal watercraft, comprises a speaker, an automated adjustment device for automatically adjusting a volume of a sound emitted from the speaker such that the volume of the sound becomes an automated sound volume value corresponding to an engine speed or a watercraft speed, and a sound volume manipulation member manipulated by a rider to change the volume of the sound emitted from the speaker with respect to the automated sound volume value; wherein when the engine speed or the watercraft speed satisfies a predetermined deceleration condition, the automated adjustment device cancels a sound volume increase amount with respect to the automated sound volume value which was caused by the rider's manipulation of the sound volume manipulation member before the predetermined deceleration condition is satisfied, and adjusts the sound volume such that the volume of the sound becomes the automated sound volume value.

Abstract: A method and apparatus for generating vibration based on sound characteristics in a mobile terminal are provided. The method includes converting audio data into an audio signal upon generation of a sound play request; determining whether to generate vibration based on a sound volume of the audio signal; setting an actuator to be driven for the audio signal among at least two actuators based on frequency distribution characteristics of the audio signal if it is determined upon determining to generate the vibration; and driving the actuator being set for the audio signal when outputting the audio signal.

Abstract: A system for estimating the occlusion effect comprises a hearing aid adapted for being set up for operation in an occlusion measurement mode. The hearing aid comprises a first transducer (9) for transforming an acoustic sound level external to a hearing aid user's ear canal into a first electrical signal. The hearing aid comprises a second transducer (10) for transforming the acoustic sound level in the occluded ear canal into a second electrical signal. The system comprises a filter bank (21, 22) for splitting the first and the second digitized electrical signals into a first and a second band split digitized electrical signal, respectively and the estimated sound pressure must be a factor ?{square root over (r)} larger than the estimated leakage when deciding if the estimated sound pressure in a given band can be applied in the calculation of the occlusion effect. The invention further provides a method for estimating the occlusion effect.

Abstract: Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, a stop band noise signal is extracted from an L+R or L?R signal produced by an FM stereo decoder. A channel quality measure is calculated based on the stop band noise signal and is used to control whether a pop suppression technique is applied to the L+R signal. The channel quality measure and the stop band noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on the L?R signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L?R and L+R signal with replacement waveforms generated via waveform extrapolation.

Abstract: A method for testing hearing aids, particularly in respect of the effect thereof on speech comprehension, is improved. The method has the following steps: a) a test system is provided, b) an audible voice signal in the form of at least one meaningless syllable is selected and presented c) the meaningless syllable and a number of further meaningless syllables are displayed on a graphical user interface in the test system, d) a heard meaningless syllable is selected from the displayed meaningless syllables by the user, e) the selection made is evaluated, and f) method steps c) to e) are repeated until a termination criterion is encountered. The audible voice signal is selected in each case on the basis of at least one selection made by the person in response to a previously presented voice signal. The method allows particularly fast and informative hearing aid tests.