Abstract: Presented herein are system(s), method(s), and apparatus for efficient background encoding/trancoding in a real time multimedia system. In one embodiment, there is presented a method for encoding/trancoding audio data. The method comprises decoding a first audio frame; executing at least one encoding task on a second audio frame, said at least one encoding task resulting in a partially encoded second audio frame, after decoding the first audio frame; decoding a third audio frame, after executing the at least the at least one encoding task; and executing at least another encoding task on the partially encoded second audio frame, after decoding the third audio frame.

Abstract: A signal processing unit for playing back an audio signal at a decelerated rate keeping pitch constant. The audio signal is at least one of a speech signal, a pure music or an audio signal which comprises of both speech and music signal. The signal processing unit comprises a plurality of bandpass filters with each of them receiving a first plurality of samples of the audio signal, a plurality of interpolators and an adder. The plurality of bandpass filters generate a second set of plurality of samples after passing the first plurality of samples of the audio signal through each of them. The plurality of bandpass filters have different pass bands, different stop bands, and a constant Q factor. The plurality of interpolators are connected to the plurality of bandpass filters and generate a third set of plurality of samples. The plurality of bandpass filters and the plurality of interpolators correspond in number.

Abstract: A signal processing unit for playing back an audio signal at an accelerated rate keeping pitch constant. The audio signal is at least one of a speech signal, a pure music or an audio signal which comprises of both speech and music signal. The signal processing unit comprises a plurality of bandpass filters with each of them receiving a first plurality of samples of the audio signal, a plurality of decimators and an adder. The plurality of bandpass filters generate a second set of plurality of samples after passing the first plurality of samples of the audio signal through each of them. The plurality of bandpass filters have different pass bands, different stop bands, and a constant Q factor. The plurality of decimators are connected to the plurality of bandpass filters and generate a third set of plurality of samples. The plurality of bandpass filters and the plurality of decimators correspond in number.

Abstract: Present herein is a low memory and MIPS efficient technique for decoding Huffman codes using multi-stage, multi-bits lookup at different levels. A binary tree is cut at levels depending on the quotient of the number of existing nodes and the number of possible nodes.

Abstract: A system and method for slowing down an audio signal while maintaining the same pitch as the original audio signal. The slowing down being done by a decoder. The method involves replicating frames of the decoded signal at a rate corresponding to the desired slow playback speed, and windowing the replicated frames to smooth out any artifacts that may result from the replication. The desired slow playback speed can be a default value predefined in the system or a value programmable by a user of the system.

Abstract: Disclosed herein is a method and system for multidimensional virtual reality audio and visual (MVR A/V) display. The MVR A/V system may be provided with an A/V capture system and an A/V display system. The MVR A/V system may also be provided with an A/V encoding system and an A/V decoding system. The MVR A/V system may also be provided with a capture storage system, a display storage system, and at least one ancillary storage system. The MVR A/V system may also be communicatively connected to a communication network, for example, the Internet. The MVR A/V method may provide capturing A/V information from a plurality of A/V receiving units, all of the A/V receiving units being focused upon at least a portion of the same zone or region. The MVR A/V method may also provide displaying the captured A/V information in a multidimensional virtual reality display system.

Abstract: A system and method for speeding up an audio signal while maintaining the same pitch as the original audio signal. The speeding up being done by a decoder. The method involves skipping frames of the decoded signal at a rate corresponding to the desired fast playback speed, and windowing the remaining frames to smooth out any artifacts that may result from skipping frames. The desired fast playback speed can be a default value predefined in the system or a value programmable by a user of the system.

Abstract: Presented herein are system(s) and method(s) for frequency domain audio speed up or slow down, while maintaining pitch. An encoded audio signal is received. Frames from the encoded audio signal are retrieved. The frames of the audio signal are transformed into a frequency domain, wherein each of said frames are associated with a plurality of initial phases, and a corresponding plurality of ending phases. The initial phases of at least one of the frames are replaced with the ending phases of another frame.

Abstract: Presented herein are systems and methods for classifying an audio signal. The audio signal is classified by calculating a plurality of linear prediction coefficients (LPC) for a portion of the audio signal; inverse filtering the portion of the audio signal with the plurality of linear prediction coefficients (LPC), thereby resulting in a residual signal; measuring the residual energy of the residual signal; and comparing the residual energy to a threshold.

Abstract: Present herein is a low memory and MIPS efficient technique for decoding Huffman codes using multi-stage, multi-bits lookup at different levels. A binary tree is cut at levels depending on the quotient of the number of existing nodes and the number of possible nodes.

Abstract: Disclosed herein is a method and system for classifying an audio signal using a sub-band energy analysis. An audio signal may be received as an input to the system for classifying an audio signal. The audio signal may be passed to a mathematical processor where the mathematical processor may perform a plurality of mathematical processes on the audio signal and calculating a ratio of energy contributable to speech and energy contributable to music. The ratio value R may be output to a comparator. The comparator may compare the calculated ratio R to a threshold value T and based upon the comparison classify the audio signal as one of speech or music.

Abstract: Disclosed herein is a method and system for classifying an audio signal. The method may be accomplished by using a low pass filter to prevent transmission of audio components having a frequency greater than a predetermined frequency. The system may also be provided with a device for selecting a further reduced number of audio components for analysis. Analysis of the audio signal may be performed by a zero point counter for counting and recording zero point transitions encountered in analysis of the audio signal. The system may also include a comparator for comparing a result of analysis to a threshold value and classifying the audio signal based upon comparison of the result of analysis and the threshold value.

Abstract: Present herein is a low memory and MIPS efficient technique for decoding Huffman codes using multi-stage, multi-bits lookup at different levels. A binary tree is cut at levels depending on the quotient of the number of existing nodes and the number of possible nodes.

Abstract: A system, method, and apparatus for calculating non-linear functions with finite order polynomials are presented herein. Use of finite order polynomials allow calculation of the non-linear functions using fixed point arithmetic operations resulting in significant cost savings.