Sony Corporation

Sony Pictures Entertainment, Inc.

Sony Electronics Inc

Other

Sony Corporation of Japan

U.S. Philips Corporation

Lucent Technologies Inc.

Laura Mercs

Paul M. Embree

Paul M Embree

James S Mercs

Paul Nigel Wood

Michael J Kohut

Casper W Barnes

Richard J Oliver

Milton L Embree

Jeffrey Mark Claar

William T Mayo

Ara James Derderian

Daniel J Minear

The present invention is a method and apparatus for allocating real-time audio data from a plurality of audio channels in a system having a first processor and a second processor. Memory banks are provided where each memory bank is accessible to the first and second processors. Subsets of the audio data are stored in the memory banks. These subsets correspond to different groups of audio channels.

Memory allocation for real-time audio processing

A method and apparatus for buffer management for digital audio recording/playback wherein a digital signal processor, memory and appropriate interfacing facilitate storage and retrieval of digital audio data samples continuously across non-contiguous areas of memory to achieve substantially seamless flow of audio, as well as precise fade-in, fade-out and cross-fading between channel events, all as required in such critical synchronizing applications as film dubbing. A GUI interface and recorder controller communication with the processor, memory and a recording media may also be included.

Buffer management system for digital audio

A decoder capable of decoding audio signals encoded using analog noise reduction techniques, particularly Dolby A, is implemented by means of digital signal processing. The encoded signal passes through one or more digital filters, producing the decoded signal. The decoded signal is fed back to drive control blocks, which determine the parameters of the digital filters. The determination of these parameters may be arranged so as to emulate closely a given analog noise reduction decoder, such as a Dolby A decoder. The one or more digital filters are relocated to the feed-forward portion of the decoder&mdash;they are located in the feedback portion of the encoder when performed using analog techniques&mdash;to overcome the problems caused by delay introduced by the digital implementation which prevent a direct conversion of the analog decoder to a digital one.

Audio noise reduction system implemented through digital signal processing

A stable and flexible synchronization system and method are disclosed. The method includes selecting to receive an external clock signal, detecting a difference between an internal clock signal and the selected external clock signal and generating a control signal representing the difference; determining if the control signal has a first value that is within a predetermined range; calculating a second value representing an absolute value of a difference between a current value of the control signal and a previous value of the control signal; determining if the value is less than a third value; resetting the output signal upon receipt of an edge of said selected external clock signal; determining if a timing difference between the output signal and the selected clock signal is substantially zero; and issuing a signal indicative of synchronization.

Flexible phase-locked loop system to indicate synchronization using difference between internal and external clock signals

The present invention overcomes the serious practical limitations inherent in presently-existing RAM-based audiovisual effects processing systems for essentially instantaneous playback of data. The method of the invention divides individual audio effects into two discrete segments, and stores these two segments in a distributed storage system. The first segment of an effect is stored in memory, allowing nearly instantaneous access and playback of this segment. The second segment is stored in a storage device with a relatively long access time. In operation, the first portion of the effect is played from memory, while the second portion is simultaneously retrieved from storage, such that the second portion is retrieved and is available for playback by the time the playing of the first segment terminates. Playback of these two segments in perfect succession effectively furnishes instantaneous playback of audio data, while reducing cost, facilitating loading of effects data, and providing further operational advantages. The disclosed method and apparatus is equally suitable for implementing essentially instantaneous playback of video or audiovisual data.

Audiovisual effects processing method and apparatus for instantaneous storage-based playback of audio data in synchronization with video data

The system and method of the present invention provides enhanced surround sound effects. In one embodiment, 90.degree. phase shift and copy of the modified head related transfer functions (HRTF) are applied to each rear sound signal. The modified HRTF for each rear signal is generated by removing the head related transfer function corresponding to the front center signal from the HRTF corresponding to the rear sound signal. This provides the audible effect of distinguishing more clearly sounds originating in front of the listener or to the rear of the listener while not limiting the perceived bandwidth of the signal. The rear signals corresponding to the first channel are inverted. The front and rear signals for each corresponding channel are then combined. The two channels, generated can be stored on media, such as film, and subsequently read and input to a surround sound decoder to generate enhanced surround sound output.

Method and apparatus for two channels of sound having directional cues

A stable and flexible phase locked loop system and method are disclosed. The system comprises a first phase frequency detector for detecting a difference between an internal clock signal and an external clock signal, and for generating a first control signal representing the difference. A first voltage controlled oscillator coupled to the first phase detector generates a first timing signal based on the first control signal. A first divider circuit coupled to the first voltage controlled oscillator divides the first timing signal by a first predetermined number to provide an output signal. A second phase frequency detector detects a difference between the first timing signal and a second timing signal, to generate a second control signal representing the difference. A second voltage controlled oscillator coupled to said second phase detector generates a third timing signal based on the second control signal. A second divider circuit coupled to the second voltage controlled oscillator divides the third timing signal by a second predetermined number to provide the second timing signal. A third divider circuit coupled to the second voltage controlled oscillator divides the third timing signal by a third predetermined number to provide the internal clock signal.

Flexible phase locked loop system

A filter for a system for processing audio samples, which dynamically vaires its length responsive to a moving average of variations in an audio input rate. The filter lengthens at substantially constant input rate variations to reduce input noise, and shortens at rapid input rate variations to enhance responsiveness.

Variable-length moving-average filter

The system and method of the present invention provides surround sound effects through a two channel system. In one embodiment, the first channel is connected to one speaker (e.g., the right speaker) of a headset and the second channel is connected to the other speaker (e.g., the left speaker) of the headset. Thus, surround sound effects, including providing effects to distinguish front and rear sound sources, are provided through headsets which isolate the sound provided to right ear and the left ear of the listener. Alternately, the signals are output to a two speaker system, enabling surround sound effects to be output through home stereo systems not necessarily configured specifically for surround sound. Modified head related transfer functions (HRTF) are applied to each rear sound signal. The modified HRTF for each rear signal is generated by removing the head related transfer function corresponding to the front center signal from the HRTF corresponding to the rear sound signal. This provides the audible effect of distinguishing more clearly sounds originating in front of the listener or to the rear of the listener while not limiting the perceived bandwidth of the signal.

Method and apparatus for electronically embedding directional cues in

A method and apparatus for sample accurate parameter update management in digital audio recording/playback wherein a digital signal processor, memory, recording controller, and appropriate interfacing facilitate storage and retrieval of digital audio data samples continuously to achieve substantially seamless flow and sample accurate audio state updated at precisely designated times, all as required in such critical synchronizing applications as film dubbing.

Sample accurate audio state update

A decoder capable of decoding audio signals encoded using analog noise reduction techniques, particularly Dolby SR, is implemented by means of digital signal processing. The typical decoding paradigm used by analog Dolby SR decoders is modified to overcome the problems caused by delay which is introduced by the digital implementation. A gain factor is factored out of the feedback transfer function and is placed in the forward signal path of each stage of the digital decoder. A single control circuit can then be used to generate coefficients which define transfer functions of each filter stage of the signal path. Linear interpolation techniques are used to limit the number of coefficient entries necessary to emulate the transfer function of an analog Dolby SR decoder. The digital decoder structure can be inverted to produce a digital encoder which is capable of implementing any noise reduction standard desired, including Dolby SR.

DSP decoder for decoding analog SR encoded audio signals

A decoder capable of decoding audio signals encoded using analog noise reduction techniques, particularly Dolby A, is implemented by means of digital signal processing. The encoded signal passes through one or more digital filters, producing the decoded signal. The decoded signal is fed back to drive control blocks, which determine the parameters of the digital filters. The determination of these parameters may be arranged so as to emulate closely a given analog noise reduction decoder, such as a Dolby A decoder. The one or more digital filters are relocated to the feed-forward portion of the decoder--they are located in the feedback portion of the encoder when performed using analog techniques--to overcome the problems caused by delay introduced by the digital implementation which prevent a direct conversion of the analog decoder to a digital one.

Audio noise reduction system implemented through digital signal

A digital surround sound decoder. The decoder uses an architecture including two signal processing chips to achieve a program that can decode audio data at sufficiently high resolution. The decoder performs in real time and is compatible with standard surround sound formats. The decoder includes software that utilizes table lookups for critical functions in the decoding process. The processing flow of the decoder's program takes advantage of the multi-function capability within the specific processors used in the design while using a minimum number of program instructions. The program implements band pass filtering, sum-difference calculations, fast-attack slow-decay integration, summation and reciprocal processing, determination of fast and slow modes, look-up table indexing, adaptive matrix processing and various other functions to generate decoded surround sound signals from encoded left and right signal inputs.

Growing community of inventors

Irvine, CA, United States of America

Paul M Embree