3567results 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.

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.

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 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.

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.

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 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.

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.