264results about How to "Improve perceived quality" patented technology

The present invention relates to methods and apparatus for increasing the perceived quality of displayed images. This is achieved in a variety of ways including the use of a plurality of device specific display characteristics when preparing images for display. It is also achieved through the monitoring of display device and/or ambient light conditions, e.g., on a periodic basis, and using the obtained information when controlling display output. Another approach to improving the perceived quality of displayed images involves the use of information relating to a specific user's ability to perceive image characteristics such as color. By customizing display output to an individual user's own physical perception capabilities and/or viewing characteristics it is possible to enhance the image quality perceived by the individual viewer as compared to embodiments which do not take into consideration individual user characteristics.

Quality settings established by an encoder are adjusted based on information associated with regions of interest (“ROIs”). For example, quantization step sizes can be reduced (to improve quality) or increased (to reduce bit rate). ROIs can be identified and quality settings can be adjusted based on input received from a user interface. An overlap setting can be determined for a portion of a picture that corresponds to an ROI overlap area. For example, an overlap setting is chosen from step sizes corresponding to a first overlapping ROI and a second overlapping ROI, or from relative reductions in step size corresponding to the first ROI and the second ROI. ROIs can be parameterized by information (e.g., using data structures) that indicates spatial dimensions of the ROIs and quality adjustment information (e.g., dead zone information, step size information, and quantization mode information).

A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.

A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal. Spectral components are synthesized having the assessed characteristics. The synthesized spectral components are integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. In one implementation, the assessed characteristic is temporal shape and noise-like spectral components are synthesized having the temporal shape of the audio signal.

A coding device having a specifying unit for specifying image region data to be processed for refreshing countermeasures in image data being coded based on that image data and a refreshing countermeasure unit for applying processing for refreshing countermeasures to image region data specified by the specifying unit when coding the image data.

An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

Disclosed herein is a conditioning system for nutritional substances. The conditioning system obtains information regarding the nutritional substance to be conditioned, the desired conditioning, and the desired properties, including nutritional content, of the conditioned nutritional substance, and dynamically controls the conditioning in response to this information optimize the organoleptic properties of the conditioned nutritional substance, while minimizing any detrimental changes to the nutritional content.

Systems and methods of transcoding video bitstreams that employ look-ahead approaches to enhance the overall perceptual quality of transcoded video information, communications, and entertainment delivered to an end user. The disclosed systems and methods of transcoding video bitstreams take into account the scene characteristics and the local coding complexity of video frames in a video sequence before performing bit allocations for the video frames, thereby significantly improving the perceptual quality of transcoded video delivered to the end user.

A post-processor 317 and method substantially for enhancing synthesised speech is disclosed. The post-processor 317 operates on a signal ex(n) derived from an excitation generator 211 typically comprising a fixed code book 203 and an adaptive code book 204, the signal ex(n) being formed from the addition of scaled outputs from the fixed code book 203 and adaptive code book 204. The post-processor operates on ex(n) by adding to it a scaled signal pv(n) derived from the adaptive code book 204. A gain or scale factor p is determined by the speech coefficients input to the excitation generator 211. The combined signal ex(n)+pv(n) is normalised by unit 316 and input to an LPC or speech synthesis filter 208, prior to being input to an audio processing unit 209.

PCT No. PCT/SE96/00128 Sec. 371 Date Aug. 4, 1997 Sec. 102(e) Date Aug. 4, 1997 PCT Filed Feb. 2, 1996 PCT Pub. No. WO96/24926 PCT Pub. Date Aug. 15, 1996A speech encoder (100) receives speech signals (S) which are encoded and transmitted on a communication channel (120). Silence in the speech is utilized by a data encoder (101) to transmit data on the speech frequency band via the channel (120). A signal classifier (103) switches between the encoders (100, 101). The speech encoder has synthesis filter (115) with state variables in a delay line, predictor adaptor (116), gain predictor (113, 114) and excitation codebook (112). The data encoder (101) has delay line with state variables stored and updated in a buffer (192). On switching (103, 102, 193) from data to speech, the buffer state variables are fed into the synthesis filter delay line via an input (144) for smooth transition in the speech encoding. Coefficient values in the synthesis filter (115) and an excitation signal (ET(1 . . . 5)) are generated. Thereby a buffer in the gain predictor (113, 114) is preset and its predictor coefficients and gain are generated. The incoming speech signal (S) newly detected is encoded (CW) by the values generated in the speech encoder (100), which is successively adapted. The receiver side has corresponding speech and data decoders.

A method of searching a plurality of Vector Quantization (VQ) codevectors for a preferred one of the VQ codevectors to be used as an output of a vector quantizer for encoding a speech signal, includes determining a quantized prediction residual vector, and calculating a corresponding unquantized prediction residual vector and the energy of the difference between these two vectors (that is, a VQ error vector). After trying each of the plurality of VQ codevectors, the codevector that minimizes the energy of the VQ error vector is selected as the output of the vector quantizer

The embodiment of the invention discloses a TD (time division) frequency planning and scrambling code planning method based on an adjacent zone priority level, comprising: an adjacent cell priority level is generated through a zone adjacent factor, a zone factor and a zone direction factor based on the base station zone attribute according to the system technical scheme; and a TD main carrier frequency is distributed according to the generated adjacent zone priority level to ensure the minimality of the frequency interferences among zones; and then, a system assigns base scrambling code blocks and scrambling codes according to the adjacent zone priority level and the frequency so as to ensure the average minimization of the mutual interference of the frequency scrambling code among zones. According to the embodiment of the invention, the calculation method of the adjacent zone priority level is more reasonable, and the setting and switching process of the adjacent zone relationship is more reasonable. Furthermore, co-frequency encounter phenomena among the adjacent zones are avoided by adopting the scheme, so that the mutual correlation of the scrambling codes among the adjacent zones is minimal, and the maximal frequency resource and scrambling code resource utilization and the minimum frequency and scrambling code interferences among zones are ensured. And the network performance indexes such as call completing rate are improved, and the client perception quality is greatly improved.

An audio signal system is provided having, in one embodiment, a magnetostrictive core of varying shapes, sizes, and permeability. The core is wound with a wire, thereby forming an inductor for producing a magnetic field when current is passed through the wire. A speaker, or other audio output device, is positioned in parallel or series with the inductor. In operation, an audio signal is received and transmitted to the speaker or other audio output device, wherein a replication of the received signal is produced. An output of the magnetostrictive inductor couples with the replicated audio signal to modify and enhance the quality of the projected signal. In yet another embodiment, digital signal processing may be used to modify the received signal as well.

A method, device, system, and computer program product expand narrowband speech signals to wideband speech signals. The method includes determining signal type information from a signal, obtaining characteristics for forming an upper band signal using the determined signal type information, determining signal noise information, using the determined signal noise information to modify the obtained characteristics for forming the upper band signal, and forming the upper band signal using the modified characteristics.