59results about How to "Improve audibility" patented technology

An apparatus for enhancing audibility of a far-end speech signal from a far-end user to a near-end user in a telephony system includes a near-end background noise signal level estimator (26) at the near-end user. A near-end speech signal level estimator (24) is also provided at the near-end user. A gain control logic (22) determines a gain (G) for amplification of the far-end speech signal based on both estimated speech and background noise signal levels.

We describe a method of blind source separation for use, for example, in a listening or hearing aid. The method processes input data from multiple microphones each receiving a mixed signal from multiple audio sources, performing independent component analysis (ICA) on the data in the time-frequency domain based on an estimation of a spectrogram of each acoustic source. The spectrograms of the sources are determined from non-negative matrix factorization (NMF) models of each source, the NMF model representing time-frequency variations in the output of an acoustic source in the time-frequency domain. The NMF and ICA models are jointly optimized, thus automatically resolving an inter-frequency permutation ambiguity.

An audio signal having at least one audio channel and associated direction parameters indicating a direction of origin of a portion of the audio channel with respect to a recording position is reconstructed to derive a reconstructed audio signal. A desired direction of origin with respect to the recording position is selected. The portion of the audio channel is modified for deriving a reconstructed portion of the reconstructed audio signal, wherein the modifying comprises increasing an intensity of the portion of the audio channel having direction parameters indicating a direction of origin close to the desired direction of origin with respect to another portion of the audio channel having direction parameters indicating a direction of origin further away from the desired direction of origin.

The invention relates to a sound system in a vehicle. The sound system comprises a first audio signal source providing a first audio signal and a second audio signal source providing a second audio signal. The sound system further comprises loudness determination unit is configured for determining a loudness of the first audio signal based on a psycho-acoustic model of a human hearing and a type information unit configured for establishing type information of the second audio signal. The sound system further comprises a control unit configured for individually selecting, for each of at least two playback zones, a mixing mode depending on the type of the second audio signal, wherein the mixing modes relate to a relation between signal output gains of the first and second audio signals. The control unit is further configured for individually mixing for playback, in each of the at least two playback zones, the first audio signal and the second audio signal based on the selected mixing mode and the determined loudness. The sound system further comprises a plurality of loudspeakers addressing individually the at least two playback zones.

An apparatus for embedding a watermark in an audio signal, the apparatus comprising:an input operable to receive the audio signal;a watermark adapting unit operable to receive the watermark from a watermark generating unit and adapt the profile of the frequency spectrum of the watermark to correspond to the profile of the frequency spectrum of the input audio signal, andwatermark embedding means operable to embed the adapted watermark in the audio signal, the watermark embedding means including a watermark gain amplifier operable to apply a gain to the watermark before the watermark is embedded in the audio signal in accordance with a gain signal generated by a watermark gain value generator, whereinthe watermark gain value generator is operable to adjust the gain applied to the watermark, the gain being determined in accordance with the presence of component of at least one peak having an amplitude above a threshold is described

With the adaptation of a hearing aid to a multitude of acoustic sources, head movements can be taken into consideration more rapidly. To this end, a position determining device (PB) for determining the current position of the head of the hearing aid wearer is provided, so that with the aid of the position of the head in a processing and separation unit (VE), the relative change in the acoustic source positions can be rapidly taken into consideration. Thus after each head movement, a renewed time-intensive adaptation is not required.

A noise reduction circuit 200 for a headphone 100 is disclosed herein. In a described embodiment, the headphone 100 includes a speaker driver 110 and the circuit 200 comprises a microphone 112 configured to convert ambient sound into a corresponding electrical ambient signal and which is disposed adjacent to the speaker driver's diaphragm. The circuit 200 further includes an active noise reduction path configured to provide active noise reduction of the ambient sound based on the corresponding electrical ambient signal and a vocal signal compensation path configured to restore attenuated signals within the vocal range of the corresponding electrical ambient signal to increase audibility of vocal signals of the ambient sound. The circuit 200 also includes a switching device 204,210 arranged to selectively deliver the corresponding electrical ambient signal to the active noise reduction path or to the vocal signal compensation path.

The invention relates to a hearing aid with a signal processing unit (14) and at least two microphones (1, 2, 3) which can be coupled together to form directional microphone systems of a different order, where microphone signals (11, 12, 13) emitted by directional microphone systems of a different order can be coupled together in a weighting dependent on the frequency of the microphone signals. The invention further relates to a method for operating a hearing aid of this type.

A mouth shield (1) for a microphone headset (4) having a rigid arcuate frame which is attachable to the mouth piece holder (3) of a microphone headset (4) by clasping means (13, 15) so it may be removably attached as desired. The shield (1) has at least one air vent aperture (2) for ease of breathing and proximal audibility. An optional removable foam insert (14) may be adhesively attached to an inner side of the shield (4) to catch saliva while speaking. The shield (1) precludes external visibility of lips while speaking and interference with audibility caused by wind and other adverse weather conditions.

An improved reverberation processor includes a technique for changing environmental parameters without causing disturbing audio artifacts. The technique includes the steps of sequentially changing the read pointers of different delay lines. Additionally, for each delay line a level control variable is ramped down prior to changing the read pointer and then ramped back up. The reverberation processor also provides means for producing and controlling a repeating echo in the reverberation decay, as well as adjusting the diffusion (or echo density) of the reverberation.

A reversing alarm for a vehicle emits an audible signal in the form of pulses of broad band sound when the vehicle is put into reverse gear. The broad band sound allows persons in the vicinity to locate the vehicle, as the human brain is able to process broad band sound to pinpoint the location of the sound source in a way which is not possible with single tone sounds. The alarm is implemented via a simple electronic circuit which uses a zener diode as a signal generator 20 to generate the broad band sound.

In accordance with an embodiment of the present invention, a dynamics processor includes a non-linear automatic gain control (AGC) responsive to an input audio signal comprised of a plurality of frequency components, each frequency component having associated therewith an amplitude, said non-linear AGC adaptive to develop a modified gain audio signal. A multiband cross-over device is responsive to the modified gain audio signal and is adaptive to generate ‘n’ number of signals, each of said ‘n’ signals having an amplitude and further having a unique frequency band associated therewith. The dynamics processor further includes ‘n’ number of processing blocks, each of which is responsive to a respective one of said ‘n’ signals for modifying the amplitude of the ‘n’ signals to develop modified ‘n’ signals; and a mixer device is responsive to said modified ‘n’ signals and adaptive to combine the same, wherein the amplitude of the plurality of frequencies associated with the audio signal is modified in real-time thereby enhancing the audibility of the audio signal.

A method of processing voice signals suitable for enhancing the speech discrimination ability of a hearing impaired person is disclosed. First, a voice signal is received, and the received voice signal is divided into a plurality of voice frames. A frequency spectrum analysis is conducted on one of the voice frames to estimate the effective bandwidth of the voice frame. Next, a frequency transposition process is performed on the voice signal so as to suit the auditory sensation bandwidth of a hearing impaired person. In addition, an energy compensation process is performed on the voice frame after performing the frequency transposition process so as to compensate the reduced energy brought by the frequency transposition process.

A method and apparatus for adding audible noise with time varying volume to audio devices are disclosed which makes the time varying volume envelope of the added audible noise proportional to the time varying volume envelope of sound for frequencies where an individual has a restricted range of perception. The method and apparatus are used to improve the audibility, speech intelligibility, and word recognition characteristics in audio devices.

A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of medical device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

A method and apparatus for adding audible noise with time varying volume to audio devices are disclosed which makes the time varying volume envelope of the added audible noise proportional to the time varying volume envelope of sound for frequencies where an individual has a restricted range of perception. The method and apparatus are used to improve the audibility, speech intelligibility, and word recognition characteristics in audio devices.

In a portable telephone, a display device, a telephone receiver, a ten-numeral entry key, a telephone transmitter, and the like are mounted on surface sides of a first housing (1) and a second housing (2), which are located on inner sides thereof when the first housing and the second housing are closed, both the first housing and the second housing being hinged to each other via a hinge portion in a freely openable / closable manner; and a speaker for producing at least a telephone call-receiving sound is provided on the surface side of any one of the first housing, the second housing, and the hinge portion. Also, a sound projection portion (12) for projecting the telephone call-receiving sound produced from the speaker is formed in a side of the portion where the speaker is provided even under such a condition that the first housing and the second housing are closed. As a result, the speaker sounds can be projected forwardly in sufficiently high levels even under such a condition that the first and second housings are opened, and furthermore, the telephone call-receiving sound can also be heard in sufficiently high levels even under such a condition that the first and second housings are closed.

The invention relates to a sound system in a vehicle. The sound system comprises a first audio signal source providing a first audio signal and a second audio signal source providing a second audio signal. The sound system further comprises loudness determination unit is configured for determining a loudness of the first audio signal based on a psycho-acoustic model of a human hearing and a type information unit configured for establishing type information of the second audio signal. The sound system further comprises a control unit configured for individually selecting, for each of at least two playback zones, a mixing mode depending on the type of the second audio signal, wherein the mixing modes relate to a relation between signal output gains of the first and second audio signals. The control unit is further configured for individually mixing for playback, in each of the at least two playback zones, the first audio signal and the second audio signal based on the selected mixing mode and the determined loudness. The sound system further comprises a plurality of loudspeakers addressing individually the at least two playback zones.

A method for a machine or group of machines to carry watermark data in an encoded audio data frame of an audio signal includes receiving the encoded audio data frame having encoded therein a portion of the audio signal. The encoded audio data frame includes a plurality of data blocks, wherein the plurality of data blocks includes, a synchronization information block, at least one encoded data block, and an error check block. The method further includes receiving modified watermark data as modified based on a masking threshold analysis of the audio signal and transforming the encoded audio data frame into a modified encoded audio data frame.

A method of programming a configurable signal processing unit of a hearing aid device comprisesProviding a frequency transposition algorithm in the hearing aid device where content from more than one upper-lying source frequency band is copied or moved into one and the same lower lying destination frequency band;Providing a number of frequency transposition configurations, each comprising a specific combination of source regions and a destination region;Providing a weight parameter that specifies the amount of gain applied to a lowered signal resulting from a given frequency transposition configuration; andProviding a prescription algorithm that selects an optimum frequency transposition configuration for an ear of a user taking an audiogram for the ear in question and an amplification capability of the hearing aid device in question into account.A purpose of the disclosure is to provide an improved audibility of high frequency sound for users with severe-to-profound hearing losses.

There is provided a method of operating a device to generate a target sound that is audible to a user of the device, the method comprising measuring background noise in an environment in which the device is located; using a perceptual loudness model to predict an audibility of the target sound to the user of the device based on the measured background noise; using the output of the perceptual loudness model to determine a gain value that is to be applied to a source signal used to generate the target sound in order to provide at least a desired level of audibility of the target sound to the user; applying the determined gain value to the source signal to produce a modified source signal; and generating the target sound using the modified source signal.

To provide a control device capable of setting a plurality of combustion parameters changing a combustion state of an internal combustion engine to appropriate values and improving a fuel consumption rate regardless of an operation state.An engine ECU 70 sets a combustion parameter (main injection timing, pilot injection timing, fuel injection pressure, turbocharging pressure, or the like) such that a center-of-gravity position of a heat generation rate becomes a constant target crank angle regardless of a load of an engine 10. In addition, the ECU 70 estimates the center-of-gravity position of a heat generation rate based on an output of an in-cylinder pressure sensor 64 and feedback-controls the combustion parameter such that the estimated center-of-gravity position of a heat generation rate becomes equal to the target crank angle.

According to the reference signal configuration and positioning method and a device, and a storage medium, each network element in the management range of the positioning server determines available TRS sending resources which can be used for sending the TRS at present, and sends the available TRS sending resources to the positioning server; the positioning server respectively receives the available TRS sending resources sent by each network element, determines TRS suggestion configuration information in combination with the available TRS sending resources of each network element, and then sends the TRS suggestion configuration information to each network element; each network element configures a TRS resource for sending the TRS based on the TRS suggestion configuration information; and the UE performs TRS detection according to the TRS suggestion configuration information sent by the positioning server or each network element so as to determine the position of the UE. Under the condition that the system complexity is not obviously increased, the existing reference signal mechanism is adjusted, the audibility of the reference signal is improved, and the positioning accuracy is improved.

A hands-free attachment for a radio telephone, comprising a transducer connectable to the receiver to generate sound waves representing sound signals received by the receiver, and electrically non-conductive means for conveying the sound waves from the transducer to an ear-piece locatable in or adjacent to the user's ear, the system preferably includes a remote microphone.