摘要:
Just in time compiled code and other data within a runtime environment may be shared between multiple applications by identifying common data objects and allowing two or more applications to access the data objects. While at least one application is accessing the objects, the objects may remain in memory. When all applications have stopped accessing an object, the object may be removed from memory. One embodiment may use a server process to manage various operations to facilitate sharing between various applications, such as identifying objects that may be removed from memory and adding newly created data to a database of sharable data.
摘要:
Just in time compiled code and other data within a runtime environment may be shared between multiple applications by identifying common data objects and allowing two or more applications to access the data objects. While at least one application is accessing the objects, the objects may remain in memory. When all applications have stopped accessing an object, the object may be removed from memory. One embodiment may use a server process to manage various operations to facilitate sharing between various applications, such as identifying objects that may be removed from memory and adding newly created data to a database of sharable data.
摘要:
A signal exhibiting redundancy, such as speech subjected to linear predictive coding, is transmitted in a reduced bandwidth by performing a linear interpolation over a number of frames. Interpolated coefficients are tested against quantized values to see if they differ by no more than a threshold. If they do not, only the last frame is sent and intermediate values are reconstructed by interpolation. If the interpolated values differ by more than the threshold from the quantized values, the number of frames for interpolation is reduced and the interpolation is repeated. This is continued until either interpolation is successful or else the next consecutive frame is sent. The required bandwidth for transmission can be varied by varying the threshold, the maximum number of frames for interpolation, the number of LPC coefficients, or a combination of these.
摘要:
A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients (βi's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.
摘要:
A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients (βi's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.
摘要:
A method of enhanced tandem communication is provided between at least a first portion of a network suitable for voice communications and a second portion of a network suitable for voice communications. During operation, two representations of an encoded signal are transmitted from the first portion of a network. The two representations comprise the encoded signal produced by a first codec and a parameter translation of the first encoded signal into an encoded signal compatible with a single common compressed voice codec (CCVC) format.
摘要:
A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.
摘要:
A signal that includes noise (301) is sampled to provide a plurality of digital information samples (303). A predetermined number of the digital information samples are grouped as a set (305). Noise suppression is performed on the signal using the following steps. One or more digital representations of silence is attached to the set, forming an extended set (401). A Fourier transform is performed on the extended set, yielding a set of frequency domain coefficients (403), at least some of which are scaled (405). An inverse Fourier transform is performed on the set of scaled frequency domain coefficients to provide a set of time domain samples (407), which are partially overlapped in time and added with a previously formed set of time domain samples (409 and 411), which result is provided with the non-overlapping time domain samples as a noise suppressed version of the signal (413).
摘要:
A method and apparatus is disclosed for improving the quality of speech samples communicated via sub-band coding utilizing adaptive bit allocation, by providing error detection only on the adaptive bit allocation information. A first error detection code, such as a cyclic redundancy check (CRC), is calculated on the bit allocation parameters in the transmitter and sent to the receiver, where a second error detection code is calculated based upon the reconstructed bit allocation parameters. The transmitted error detection code is then used to determine if the received bit allocation information is correct, and if not, the frame of speech data is discarded. By protecting only the bit allocation information, additional speech frames may be salvaged from the error-prone channel, thus further increasing speech intelligibility.
摘要:
An improved method and apparatus for measuring the strength of a radio frequency (RF) signal subject to Rayleigh fading is described. The strength of the RF signal is sampled two or more times during a predetermined time interval and the sampled signal strength having the largest magnitude is selected. The selected signal strength is a reasonably accurate measure of the signal strength since the true average strength of a Rayleigh fading signal is close to its peak signal strength. The inventive method and apparatus is particularly well adapted for use in a scanning receiver located in a base station radio of a cellular radiotelephone communications system. The scanning receiver includes an antenna selector for selecting one of a plurality of sector antennas, an RF signal receiver tunable to a plurality of different RF signal frequencies, an analog-to-digital converter for converting RF signal strength samples to a binary data signal, and a microprocessor together with peripheral devices for controlling the operation of the antenna selector, RF signal receiver and analog-to-digital converter.