3608results about How to "Improve sound quality" patented technology

The invention relates to mobile audio playback devices and methods of its operation for enhanced music and sound experience for mobile devices. To provide stereo and surround sound, the present invention provides an audio playback device with a receiving means for receiving multi-channel audio data, obtaining means connected to the receiving means, for obtaining first channel audio data from the multi-channel audio data for playback on the audio playback device, and playback means having at least one loudspeaker, and being connected to the obtaining means for outputting the first channel audio data. The other not selected audio channels or all received audio data may be transferred to other terminals for playback or may be discarded.

A management and control unit for a network-based conferencing system has an interface for outputting control signals to a platform for establishing audio connections across a network between users. Another interface receives control signals from a platform for providing a graphical user interface to a user, for use in controlling the conferencing system. The unit has access to a database for maintaining management data relating to one or more existing conferences. The management and control unit receives control signals input by a user at the graphical user interface in respect of an audio conference, and outputs control signals to the platform for establishing audio connections. An audio conference can thus be established between the user and at least two other users over the network. Management data can be transmitted to the graphical user interface during a conference for use in managing the conference.

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 system and method for providing thermal gender recognition of people in any place of entertainment, leisure and / or recreation. The system utilizes infrared thermal imaging cameras and enables a prospective attendee determine in advance from a remote location where the prospective attendee would like to be. The system includes at least one computer system capable of communicating with a server over a network for retrieval, processing, and transmission of captured image data of each person in view of the infrared thermal imaging cameras. A website is configured to receive the image data of each person from the at least one computer system and implement the image data in a suitable format for viewing of the icons in various locations of the place by the prospective attendee from the remote location.

A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.

Patient may operate a CPAP system to deliver appropriate airway pressure home. A patient's apnea problem can be diagnosed at home without supervision with a CPAP device which delivers a continuously minimum appropriate pressure for substantially the entire period of therapy.

The present invention is directed to a call admission controller that is operable to: (a) determine at least one of (i) a bandwidth utilization level for a first path including a first link; (ii) an available bandwidth level for the first path; and (iii) one or more Quality of Service or QoS metrics for the first path; (b) compare the at least one of (i) a bandwidth utilization level; (ii) an available bandwidth level; and (iii) one or more Quality of Service or QoS metrics to one or more selected thresholds to determine whether a new live voice communication may be set up with a first selected codec; and (iii) when a new live voice communication may not be set up with the first selected codec, perform at least one of the following operations: (i) select a second different codec from among a plurality of possible codecs for the new live voice communication, wherein the second codec has a lower bit rate than the first codec; (ii) change an existing live voice communication from the first codec to the second codec; and (iii) redirect the new live voice communication from the first path to a second different path, wherein the second path does not include the first link.

An audio system, an audio outputting method, and a speaker apparatus are disclosed. The audio system includes a plurality of speaker modules connected to each other, a detection module configured to detect information of the plurality of speaker modules and user information, and a home control module configured to receive an audio signal, process the received audio signal based on the information of the plurality of speaker modules and the user information, and transmit the processed audio signal to the plurality of speaker modules.

According to the first aspect of the invention, a hearing instrument with at least one microphone and signal processing comprises at least two receivers having a different frequency response. At least a first one of the receivers is placed outside the ear canal, for example in a behind-the-ear component. According to the second aspect of the invention, a hearing instrument comprising a behind-the-ear component and an external component for being placed in the user's ear canal or in the user's ear as well as a connection link between the behind-the-ear component and the external component is provided, where the connection link is reversibly pluggable to the behind-the-ear component and / or the in-the-ear-canal component and has a length that is reversibly adjustable. The hearing instrument also comprises fixation means for reversibly fixing the adjusted length of the connection link. A hearing instrument according to the third aspect comprises a fixation means separate from the in-the-ear-canal component. The fixation means may be positioned in the ear canal and rest therein. The in-the-ear-canal component may be connected to the fixation means and detached therefrom when the same is already in the ear canal. A hearing instrument according to the fourth aspect of the invention comprises an in-the-ear-canal component to be placed in a user's ear canal and fixation means for fixing it in the ear canal. The fixation means comprise an outer shell which is shaped to fit (i.e. custom shaped to fit the specific user's ear geometry) in the user's ear canal and a mounting structure for holding the in-the-ear-canal component. The outer shell is preferably resilient, i.e. has an elasticity allowing temporal deformation.

An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

A system and method transmit a data stream from a source to a destination over a communication channel. A transmitter includes devices for processing inputs to assemble data packets for the data stream, and a multiplexer for assembling a data frame to be transmitted over the communication channel, in which each data frame has at least one fixed slot. The multiplexer sets at least one freely allocatable time slot in each data frame. Retransmission control devices connected to the multiplexer retransmit a specific data packet which is not properly received by the destination, using one of the freely allocatable slots.

A system for use with an audio reproduction system that corrects for distortion caused by the system as well as any hearing impairment suffered a listener. Test signals are introduced into the input of the system to produce test sounds that are perceptible by the listener. Using a pushbutton, the listener indicates when a test signal of progressively increasing volume reaches an audible level. The resulting measured values of the listener's threshold of hearing at different frequencies is compared with comparable data for a normal listener to generate correction values that are used to program an equalizer which compensates for not only the listener's hearing impairments but also any distortion produced by system components or room acoustics.

A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal.

The present invention relates to a solution for improving the perceived sound quality of a decoded acoustic signal. The improvement is accomplished by means of extending the spectrum of a received narrow-band acoustic signal (aNB). According to the invention, a wide-band acoustic signal (aWB) is produced by extracting at least one essential attribute (zNB) from the narrow-band acoustic signal (aNB). Parameters, e.g. representing signal energies, with respect to wide-band frequency components outside the spectrum (ANB) of the narrow-band acoustic signal (aNB) are estimated based on the at least one essential attribute (zNB). This estimation involves allocating a parameter value to a wide-band frequency component, based on a corresponding confidence level. For instance, a relatively high parameter value is allowed to be allocated to a frequency component if it has a comparatively high degree certainty. In contrast, a relatively low parameter value is only allowed to be allocated to a frequency component if it is associated with a comparatively low degree certainty.

An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

An acoustic data collection system that uses an antenna array aboard a powered unmanned and remotely controlled parafoil. The antenna array has multiple microphone elements, whose outputs are combined to provide directionality to the data acquisition.

The present invention relates to a display unit for a refrigerator for improving sound quality. A display having a TV receiving function is installed at a front surface of a refrigerator door. A mounting panel with a plurality of speaker holes formed therein is also installed around an edge of the display. A speaker is mounted to a rear surface of the mounting panel on a portion where the speaker holes are formed. Further, a resonance box is installed to enclose the speaker such that low and middle frequency sound can be transmitted forward to the outside. A sound transmission tube that communicates with the speaker hole is formed at a front portion of the resonance box. In addition, a sound absorbing material is covered on an inner rear surface of the resonance box that faces the rear of the speaker.

An audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation.

A scalable audio coding / decoding method and apparatus are provided. The coding method includes the steps of (a) signal-processing input audio signals and quantizing the same for each predetermined coding band; (b) coding the quantized data corresponding to the base layer within a predetermined layer size; (c) coding the quantized data corresponding to the next enhancement layer of the coded base layer and the remaining quantized data uncoded and belonging to the enhancement layer, within a predetermined layer size; and (d) sequentially performing the layer coding steps for all layers, wherein the steps (b), (c) and (d) each comprise the steps of: (i) obtaining gamut bit allocation information representing the number of bits of the quantized data corresponding to the respective subbands belonging to a layer to be coded; (ii) obtaining the number of bits allocated to the respective subbands within each subband size of the layers; (iii) generating an index representing the presence of quantized data for predetermined frequency components forming the subbands for the quantized data corresponding to the number of allocated bits; and (iv) generating bitstreams by coding the quantized data corresponding to the gamut bit allocation information, quantization step size, index and number of bits allocated to the respective subbands, by a predetermined coding method.

A process for combining audio channels combines the audio channels to produce a combined audio channel and dynamically applies one or more of time, phase, and amplitude or power adjustments to the channels, to the combined channel, or to both the channels and the combined channel. One or more of the adjustments are controlled at least in part by a measure of auditory events in one or more of the channels and / or the combined channel. Applications include the presentation of multichannel audio in cinemas and vehicles. Not only methods, but also corresponding computer program implementations and apparatus implementations are included.

To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

The invention provides a Chinese speech synthesis method based on phonemes and rhythm structures. The method is divided into a training stage and a synthesis stage, and comprises the steps of extracting therhythm structure features from the rhythm labeling information in a to-be-processed text according to the linguistic knowledge, and training a rhythm model based on the rhythm structure features; preprocessing the to-be-processed text and an audio to obtain a pinyin sequence containing the rhythm information and the corresponding acoustic features, then training an acoustic model, and deploying the trained rhythm model and acoustic model to a background; obtaining a text containing the rhythm information corresponding to the input text according to the rhythm model; converting into a pinyin sequence with rhythm information, and inputting into the acoustic model to obtain a linear frequency spectrum; and converting the linear frequency spectrum into the audio. According to the Chinesespeech synthesis method, the synthesized voice can be more natural, and especially for the longer clauses, the pause position can be determined. In addition, the model is deployed to the background at the synthesis stage, and the model loading time is saved, so that the voice synthesis speed can be increased.

Errors in pitch (frequency) allocation within a cochlear implant are corrected in order to provide a significant and profound improvement in the quality of sound perceived by the cochlear implant user. In one embodiment, the user is stimulated with a reference signal, e.g., the tone “A” (440 Hz) and then the user is stimulated with a probe signal, separated from the reference signal by an octave, e.g., high “A” (880 Hz). The user adjusts the location where the probe signal is applied, using current steering, until the pitch of the probe signal, as perceived by the user, matches the pitch of the reference signal, as perceived by the user. In this manner, the user maps frequencies to stimulation locations in order to tune his or her implant system to his or her unique cochlea.

An audio signal coding apparatus includes a first-stage encoder for quantizing the time-to-frequency transformed audio signal and second-and-subsequent-stages of encoders each for quantizing a quantization error output from the previous-stage encoder A characteristic decision unit is provided which decides the frequency band of an audio signal to be quantized by each encoder of multiple-stage encoders, and a coding band control unit receives the frequency band decided by the characteristic decision unit and the time-to-frequency transformed audio signal, decides the order of connecting the respective encoders, and transforms the quantization bands of the encoders and the connecting order to code sequences. Therefore, it is possible to provide an audio signal coding apparatus performing adaptive scalable coding, which exhibits sufficient performance when coding various audio signals.

The invention discloses a conversion method for sound of a speaker. The method comprises a training stage and a conversion stage, wherein the training stage comprises the steps of respectively extracting a fundamental frequency characteristic, a speaker characteristic and a content characteristic from training voice signals of a source speaker and a target speaker, constructing a fundamental frequency conversion function according to the fundamental frequency characteristic, and constructing a speaker conversion function according to the speaker characteristic. The conversion stage comprises the steps of extracting a fundamental frequency characteristic and a spectrum characteristic from a voice signal to be converted of the source speaker, using the fundamental frequency conversion function and the speaker conversion function obtained in the training stage to convert the fundamental frequency characteristic and the speaker characteristic extracted from the voice signal to be converted, obtaining the converted fundamental frequency characteristic and the speaker characteristic, and synthesizing voices of the target speaker according to the obtained converted fundamental frequency characteristic, the speaker characteristic and the content characteristic in the voice signal to be converted. The method is easy to realize, and the converted sound quality and similarity are higher.

A scalable audio coding / decoding method and apparatus are provided. The coding method includes the steps of (a) signal-processing input audio signals and quantizing the same for each predetermined coding band; (b) coding the quantized data corresponding to the base layer within a predetermined layer size; (c) coding the quantized data corresponding to the next enhancement layer of the coded base layer and the remaining quantized data uncoded and belonging to the enhancement layer, within a predetermined layer size; and (d) sequentially performing the layer coding steps for all layers, wherein the steps (b), (c) and (d) each comprise the steps of: (i) obtaining gamut bit allocation information representing the number of bits of the quantized data corresponding to the respective subbands belonging to a layer to be coded; (ii) obtaining the number of bits allocated to the respective subbands within each subband size of the layers; (iii) generating an index representing the presence of quantized data for predetermined frequency components forming the subbands for the quantized data corresponding to the number of allocated bits; and (iv) generating bitstreams by coding the quantized data corresponding to the gamut bit allocation information, quantization step size, index and number of bits allocated to the respective subbands, by a predetermined coding method.