Abstract: The method and system of present invention sequences audio post-processing algorithms to simulate live or theater sound. An audio signal is selectively post-processed according to equipment availability and listener preferences. Downmixing or Prologic algorithms are applied to a signal arriving at sound system. A listener inputs their speaker configuration to a player console. Desired post-processing effects are likewise indicated to the console. For instance, if surround sound equipment is both available and selected, then surround portions of the audio signal are parsed to surround speakers. Bass management techniques then transfer low frequency channels of the signal to compatible speakers. VES or DCS algorithms further manipulate the surround portion of the signal to create an illusion of immersion, and a center channel equalizer balances the signal playback. Alternatively, the post-processed signal is transmitted to a headphone set.

Abstract: A system and method for accurately calculating a mathematical power function in an electronic device may include an application program that is configured to calculate a direct estimate of power function value for the mathematical power function during a direct linear interpolations procedure. The application program may also calculate an indirect estimate of power function value for a complement power function during an indirect linear interpolation procedure. The application program may then perform a final function-estimate calculation procedure to accurately produce a final estimated power function value from the foregoing direct estimate of power function value and indirect estimate of power function value.

Abstract: A method, apparatus and program product facilitates the sharing of memory resources between exclusive audio post-processes. A program identifies post-processing applications that execute at different instances and assigns to them a common memory block. An audio packet arrives at a digital signal processor (DSP). The DSP associates a frame of the packet with a post-process. The DSP buffers the frame to a memory block that corresponds to the post-process. Upon releasing the buffered frame, the DSP prepares the memory block for use with a second post-process and frame.

Abstract: The method and system of present invention sequences audio post-processing algorithms to simulate live or theater sound. An audio signal is selectively post-processed according to equipment availability and listener preferences. Downmixing or Prologic algorithms are applied to a signal arriving at sound system. A listener inputs their speaker configuration to a player console. Desired post-processing effects are likewise indicated to the console. For instance, if surround sound equipment is both available and selected, then surround portions of the audio signal are parsed to surround speakers. Bass management techniques then transfer low frequency channels of the signal to compatible speakers. VES or DCS algorithms further manipulate the surround portion of the signal to create an illusion of immersion, and a center channel equalizer balances the signal playback. Alternatively, the post-processed signal is transmitted to a headphone set.

Abstract: A method, apparatus and program product facilitates the sharing of memory resources between exclusive audio post-processes. A program identifies post-processing applications that execute at different instances and assigns to them a common memory block. An audio packet arrives at a digital signal processor (DSP). The DSP associates a frame of the packet with a post-process. The DSP buffers the frame to a memory block that corresponds to the post-process. Upon releasing the buffered frame, the DSP prepares the memory block for use with a second post-process and frame.

Abstract: Method and apparatus for reducing the occurrence of overflow in an IIR filter. The feedback state of a filter is stored with double precision and scaled-down to increase range.

Abstract: A bitstream is sampled at an interval predetermined as a function of available processing power and memory storage capacity. The sampled bits are evaluated against a first layer Huffman decoding subtable. The subtable obtains the address of a second layer subtable from an indexed location identified by the sampled bits. A second bit sample is parsed to the designated second layer subtable, which outputs either an address for a third layer subtable, or a symbol and an associated code length.

Abstract: A method and apparatus for calibrating antenna array systems comprising of a portable transponder (118) and a calibration processor (124). Calibration processor (124) generates calibration transmit signals which it transmits using transmit signal processor (104) and antenna array (111). Transponder (118) echoes these signals back and they are acquired by calibration processor (124) through antenna array (111) and receive signal processor (106). Calibration processor (124) processes the received signals to calculate antenna calibration vector (102).