Abstract: The invention uses digital signal processing (DSP) techniques to synchronize an audio encoding process with a video synchronization signal. Namely, the encoder parameters of a DSP microchip are preset according to characteristics of an audio frame. A buffer temporarily stores the audio frame prior to sending it to an encoder. The buffer then transfers the frame in response to receiving a video synchronization signal in conjunction with authorization from a microprocessor. As such, the encoding sequence of the audio frame coincides with the video synchronization signal. Since the corresponding video frame is already slaved to the video synchronization signal, the audio samples are effectively processed in sequence with the video data. Prior to outputting the encoded audio frame to a multiplexor, the encoder sends a value to the microprocessor representing the difference between the end of the encoded audio frame and a second video synchronization signal.

Abstract: A system for improved digital data compression in an audio encoder. A threshold is established which depends on the bit rate of the input data. A determination is made whether the bit rate is above or below the established threshold. A masking index is calculated for the input data according to a first formula if the input data is being transmitted at a rate at or below the threshold. A second formula is used to calculate the masking index if the input data is being transmitted at a rate above the threshold. The masking index is used to generate a masking threshold, and data deemed insignificant relative to the masking threshold is ignored. In the preferred embodiment of the present invention, a psycho-acoustic modeler, which is included in the encoding section of an encoding/decoding (CODEC) circuit, is used to determine a masking index. The masking index is then used to generate a masking threshold.

Abstract: Digital signal processing systems and methods for smoothing out an audio signal having an abrupt transition in volume. A DSP in an audio visual recording device is configured to smooth out discontinuous volume transitions in an audio signal during encoding. Such abrupt, discontinuous changes can occur when a user instructs the recording device to pause and resume during recording or when the user changes channels during recording. The DSP applies a transform that linearly increases or decreases the volume over one or more audio frames from the first volume level to the next volume level after the abrupt change.

Abstract: The present invention relates to a system and method which serves as a refinement in the criteria used to improve the performance of audio signal processing systems. More specifically, the present invention provides a method by which the frequency and magnitude of artifacts added to audio signal data in an encoder device can be reduced. The encoding device through which the audio signal passes includes a filter bank for filtering source audio data to produce frequency sub-bands, a psycho-acoustic modeler for calculating signal to masking ratios from the frequency sub-bands of the source audio data, and a bit allocator for assigning for using the signal to masking ratios to assign a finite number of bits to represent the frequency sub-bands. In the absence of a significant event, the bit allocator performs a pre-bit allocation procedure to prevent artifacts or discontinuities in the encoded audio data.

Abstract: The invention uses digital signal processing (DSP) techniques to synchronize an audio encoding process with a video synchronization signal. Namely, the encoder parameters of a DSP microchip are preset according to characteristics of an audio frame. A buffer temporarily stores the audio frame prior to sending it to an encoder. The buffer then transfers the frame in response to receiving a video synchronization signal in conjunction with authorization from a microprocessor. As such, the encoding sequence of the audio frame coincides with the video synchronization signal. Since the corresponding video frame is already slaved to the video synchronization signal, the audio samples are effectively processed in sequence with the video data. Prior to outputting the encoded audio frame to a multiplexor, the encoder sends a value to the microprocessor representing the difference between the end of the encoded audio frame and a second video synchronization signal.

Abstract: A system comprises a refined psycho-acoustic modeler for efficient perceptive encoding compression of digital audio. Perceptive encoding uses experimentally derived knowledge of human hearing to compress audio by deleting data corresponding to sounds which will not be perceived by the human ear. A psycho-acoustic modeler produces masking information that is used in the perceptive encoding system to specify which amplitudes and frequencies may be safely ignored without compromising sound fidelity. The present invention includes a system and method for efficiently implementing a masking function in a psycho-acoustic modeler in digital audio perceptive encoding. In the preferred embodiment, the present invention comprises a non-logarithmically based representation of individual masking functions utilizing minimally-sized look-up tables.

Abstract: An apparatus and method for implementing a variable-speed audio data playback system comprises a processor that truncates audio data into frames and then performs a series of synchronized crossfades to discard excess audio data that is unneccessary during variable-speed playback. During the synchronized crossfade, the processor stores a source frame and a destination frame into memory. The processor then locates a maximum crosscorrelation point between the source frame and the destination frame and performs a synchronized crossfade that starts at the maximum crosscorrelation point to thereby produce high-quality reproduced sound to a system user during a variable-speed playback mode.

Abstract: A system comprises a refined psycho-acoustic modeler for efficient perceptive encoding compression of digital audio. Perceptive encoding uses experimentally derived knowledge of human hearing to compress audio by deleting data corresponding to sounds which will not be perceived by the human ear. A psycho-acoustic modeler produces masking information that is used in the perceptive encoding system to specify which amplitudes and frequencies may be safely ignored without compromising sound fidelity. The present invention includes a system and method for efficiently implementing a masking function in a psycho-acoustic modeler in digital audio perceptive encoding. In the preferred embodiment, the present invention comprises a non-logarithmically based representation of individual masking functions utilizing minimally-sized look-up tables.

Abstract: A system comprises a refined psycho-acoustic modeler for efficient perceptive encoding compression of digital audio. Perceptive encoding uses experimentally derived knowledge of human hearing to compress audio by deleting data corresponding to sounds which will not be perceived by the human ear. A psycho-acoustic modeler produces masking information that is used in the perceptive encoding system to specify which amplitudes and frequencies may be safely ignored without compromising sound fidelity. The present invention includes a system and method for efficiently implementing a masking function in a psycho-acoustic modeler in digital audio perceptive encoding. In the preferred embodiment, the present invention comprises a non-logarithmically based representation of individual masking functions utilizing minimally-sized look-up tables.

Abstract: A system comprises a refined psycho-acoustic modeler for efficient perceptive encoding compression of digital audio. Perceptive encoding uses experimentally derived knowledge of human hearing to compress audio by deleting data corresponding to sounds which will not be perceived by the human ear. A psycho-acoustic modeler produces masking information that is used in the perceptive encoding system to specify which amplitudes and frequencies may be safely ignored without compromising sound fidelity. The present invention includes a refined approximation to the experimentally derived individual masking spread function, which allows superior performance when used to calculate the overall amplitudes and frequencies that may be ignored. The present invention also includes an enhanced tonal component determiner, which allows for the more accurate identification of significant tonal components.