Microsoft Technology Licensing, LLC

At&t Corp.

Immersion Networks, Inc.

At&t Intellectual Property Ii, L.p.

Lucent Technologies Inc.

At&t Bell Laboratories

Dts, Inc.

Bell Telephone Laboratories

Other

Fraunhofer-gesellschaft Zur Foerderung Der Angewandten Forschung E.v.

American Telephone & Telegraph Co., At&t Bell Laboratories

At&t Ipm Corp.

James David Johnston

Shyh-Shiaw Kuo

Schuyler Reynier Quackenbush

Stephen Daniel White

King Wei Hor

Barry M Genova

William Turin

Serge Smirnov

Robert J Safranek

Sean Matthew Dorward

Jean-Marc Jot

Chris Messer

Arun N Netravali

Kim N Matthews

Deepen Sinha

Vij Rajarajan

Scott C Knauer

Eric D Petajan

Zoran Fejzo

Casey Kiernan

Peter H Westerink

Mel MacMahon

Kishnan Nedungadi

Kevin Hodge

Cary Lane Rohwer

Stephen Roger Hastings

Lauren Beth Gallagher

Tom Martino

Annette B Hall

Nambirajan Seshadri

Zhengyou Zhang

Naveen Thumpudi

Nuggehally Sampath Jayant

James L Flanagan

Eric R Wagner

Anandha K Ganesan

Brijesh D Bhatia

Sudheer Sirivara

David L Neuhoff

Thrasyvoulos N Pappas

Andrew Hayes

Joseph L Hall

John Bregar

Qunli Li

Audrius Zimnickas

Jin Li

Martin Dietz

Wei-Ge Chen

Robert George Atkinson

Wei-ge Chen

Jay Gordon Wilpon

Carl-Erik Wilhelm Sundberg

Gary Wayne Elko

Karlheinz Brandenburg

Lawrence Richard Rabiner

Aaron Edward Rosenberg

Kirk O Olynyk

Michael W Van Flandern

Heinz Gerhauser

Bodo Teichmann

Jonathan B Allen

David Joel Goodman

J&uuml;rgen Herre

James A Reeds

Kevin Michael Schofield

Sergey Smirnov

Uwe Gbur

Kenneth Lane Thompson

Tyler Gleghorn

Mark R Sydorenko

Audrius Zimnicks

Anibal J Ferreira

John Peter Princen

A Michael Noll

Wai Yip Chan

Lori F Lamel

William Tom Blank

A method for loudness equalization is provided that includes receiving input loudness data at an audio processing system. Converting gain data of the input loudness data to a linear scale at the audio processing system. Determining a reciprocal of a gain-linear loudness value as a function of the converted gain data using the audio processing system. Determining a compression ratio using the audio processing system. Performing temporal smoothing and look ahead processing using the audio processing system. Outputting gain data as a function of the temporal smoothing and look ahead processing using the audio processing system.

Loudness equalization system

An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events.

System and method for processing audio data into a plurality of frequency components

An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events. The encoder operable to transform the error signal into high resolution frequency components using the MDCT block sizes, quantize the scale factors and frequency components, and encode the quantized lines, block sizes, and quantized scale factors for inclusion in the bitstream.

A method for encoding an audio signal, comprising using one or more algorithms operating on a processor to filter the audio signal into two output signals, wherein each output signal has a sampling rate that is equal to a sampling rate of the audio signal, and wherein one of the output signals includes high frequency data. Using one or more algorithms operating on the processor to window the high frequency data by selecting a set of the high frequency data. Using one or more algorithms operating on the processor to determine a set of linear predictive coding (LPC) coefficients for the windowed data. Using one or more algorithms operating on the processor to generate energy scale values for the windowed data. Using one or more algorithms operating on the processor to generate an encoded high frequency bitstream.

Systems and methods for processing high frequency audio signal

System and method for processing audio data

A decoder generates decoded signals based on quantized signals. The decoder includes an inverse quantizer and a predictor circuit. The quantized signals are generated in an encoder by low-pass filtering an input signal and encoding the filtered signal using adaptive differential pulse code modulation. The predictor circuit has filter coefficients based on a frequency response of the low-pass filter used to filter the input signal.

Adaptive audio decoder system, method and article

An encoder generates quantized signal words based on a difference signal. The encoder includes an adaptive quantizer. A step size applied by the adaptive quantizer is generated in a feedback loop and based on a loading factor and quantized signal words generated by the adaptive quantizer. The encoder includes coding circuitry which generates code words based on quantized signal words generated by the adaptive quantizer. The coding circuitry generates an escape code in response to a quantized signal word not being associated with a corresponding coding code word.

Adaptive audio codec system, method and article

An encoder includes a low-pass filter to filter input audio signals. The low-pass filter has fixed filter coefficients. The encoder generates quantized signals based on a difference signal. The encoder includes an adaptive quantizer and a decoder to generate feedback signals. The decoder has an inverse quantizer and a predictor. The predictor has fixed control parameters which are based on a frequency response of the low-pass filter. The predictor may include a finite impulse response filter having fixed filter coefficients. The decoder may include an adaptive noise shaping filter coupled between the low-pass filter and the encoder. The adaptive noise shaping filter flattens signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.

Adaptive audio encoder system, method and article

A codec operable to process audio data and related data. The codec further operable to receive at least one of an audio, audio auxiliary, program configuration, and data signals from a program source, the audio signals including at least one of single channel audio and multi-channel audio signals, audio auxiliary signals including spatial and motion data and environmental characteristics, the data signals including program related data. The codec further operable to generate a non-transitory encoded bitstream, wherein the bitstream includes at least one of synchronization command data and at least one of a program command data, audio channel data, audio auxiliary data, program content data, and an end of stream data, wherein the encoded bitstream includes an identifier for defining packet type for each data component. The synchronization command data includes a stream start flag defining an entry point for decoding the bitstream and further provides sample rate for the encoded bitstream.

A sub-band coder operable to process audio samples for use in a digital encoder. The sub-band coder comprising application code instructions executable on a processor configured to cause the coder to filter the audio samples into a plurality of frequency band components using at least one Pseudo-Quadrature Mirror Filter (PQMF) and modulate the plurality of frequency band components into a plurality of quantized band values using a pulse code modulation technique. The application code instructions further operable to cause the coder to decode the plurality of quantized band values into an approximation signal using an inverse pulse code modulation technique and at least one Inverse Pseudo-Quadrature Mirror Filter (IPQMF). The application code instructions operable to cause the coder generates an output for use by the digital encoder that includes the plurality of quantized band values, the approximation signal, and a plurality of encoded quantized band values.

A decoder operable to decode audio signals. The decoder operable to receive an encoded bitstream that includes bitstream synchronization command data and program command data and process the encoded bitstream and identify within the bitstream the synchronization command data. The decoder further operable to decode the program command packet and at least one program related channel data using information provided in the synchronization command data and decode program related channel data using information provided in the program command data.

The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a 'short' block or three filters for a 'long' block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

Effective deployment of Temporal Noise Shaping (TNS) filters

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Morristown, NJ, United States of America

James David Johnston