Lucent Technologies Inc.

Agere Systems Inc.

Sirius Xm Radio Inc.

Agere Systems Guardian Corp.

Other

At+t Corp.

Ibiquity Digital Corporation

Sirus Xm Radio Inc.

Mosaid Technologies Corporation

Deepen Sinha

Carl-Erik Wilhelm Sundberg

Christopher Ward

Raymond Lowe

Mark Kalman

James David Johnston

Paul D Marko

Peter Kroon

Jerry Nicholas Laneman

Hariom Aggrawal

Hui-Ling Lou

Jonathan B Allen

Mark R Sydorenko

William J Casey, Iii

Erumbi Vallabhan Harinarayanan

James Walter Tracey

Carl-Eric Wilhelm Sundberg

David Mansour

Masoud Alghoniemy

Lin Lin

Alex Cabanilla

Anibal J S Ferreira

Nicholas G Karter

Systems and methods are presented for efficient cross-fading of compressed domain information streams on a user/client device. Exemplary systems may provide cross-fade between AAC/Enhanced AAC Plus information streams, between MP3 information streams, or between information streams of unmatched formats. These systems are distinguished in that cross-fade is directly applied to compressed bitstreams so a single decode operation is performed on the resulting bitstream. Thus, a set of frames from each input stream associated with the time interval in which a cross fade is decoded, and combined and recoded with a cross fade or other effect now in the compressed bitstream. Once sent through the client device's decoder, the user hears the transitional effect. The only input data that is decoded and processed is that associated with the portion of each stream used the crossfade, blend or other interstitial, and thus the vast majority of input streams are left compressed.

Systems and methods for implementing efficient cross-fading between compressed audio streams

Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPlus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented. Thus, in exemplary embodiments of the present invention a set of frames from each input stream associated with the time interval in which a cross fade is decoded, and combined and recoded with a cross fade or other effect now in the compressed bitstream. Once sent through the client device's decoder, the user hears the transitional effect. The only input data that is decoded and processed is that associated with the portion of each stream used in the crossfade, blend or other interstitial, and thus the vast majority of the input streams are left compressed.

Systems and methods for increasing transmission bandwidth efficiency by the analysis and synthesis of the ultimate components of transmitted content are presented. To implement such a system, a dictionary or database of elemental codewords can be generated from a set of audio clips. Using such a database, a given arbitrary song or other audio file can be expressed as a series of such codewords, where each given codeword in the series is a compressed audio packet that can be used as is, or, for example, can be tagged to be modified to better match the corresponding portion of the original audio file. Each codeword in the database has an index number or unique identifier. For a relatively small number of bits used in a unique ID, e.g. 27-30, several hundreds of millions of codewords can be uniquely identified. By providing the database of codewords to receivers of a broadcast or content delivery system in advance, instead of broadcasting or streaming the actual compressed audio signal, all that need be transmitted is the series of identifiers along with any modification instructions to the identified codewords. After reception, intelligence on the receiver having access to a locally stored copy of the dictionary can reconstruct the original audio clip by accessing the codewords via the received IDs, modify them as instructed by the modification instructions, further modify the codewords either individually or in groups using the audio profile of the original audio file (also sent by the encoder) and play back a generated sequence of phase corrected codewords and modified codewords as instructed. In exemplary embodiments of the present invention, such modification can extend into neighboring codewords, and can utilize either or both (i) cross correlation based time alignment and (ii) phase continuity between harmonics, to achieve higher fidelity to the original audio clip.

System and method for increasing transmission bandwidth efficiency ('EBT2')

Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPIus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented. Thus, in exemplary embodiments of the present invention a set of frames from each input stream associated with the time interval in which a cross fade is decoded, and combined and recoded with a cross fade or other effect now in the compressed bitstream. Once sent through the client device's decoder, the user hears the transitional effect. The only input data that is decoded and processed is that associated with the portion of each stream used in the crossfade, blend or other interstitial, and thus the vast majority of the input streams are left compressed.

A novel bandwidth extension technique allows information to be encoded and decoded using a fractal self similarity model or an accurate spectral replacement model, or both. Also a multi-band temporal amplitude coding technique, useful as an enhancement to any coding/decoding technique, helps with accurate reconstruction of the temporal envelope and employs a utility filterbank. A perceptual coder using a comodulation masking release model, operating typically with more conventional perceptual coders, makes the perceptual model more accurate and hence increases the efficiency of the overall perceptual coder.

Method and apparatus for audio encoding and decoding using wideband psychoacoustic modeling and bandwidth extension

An efficient coding scheme with higher audio bandwidth and/or better audio quality at lower bitrates, wherein the scheme eliminates long-term and short-term frequency domain correlation in a signal via frequency domain predictors. The coding scheme compresses information consisting of coded low frequency components as well as a parametric representation for the high frequency components based on a non-linear model. Additionally, by working on the frequency domain representations of the signal (such as the MDCT representation which is naturally available to a PAC encoder and decoder), low pass and high pass signal components are easily obtained by windowing the appropriate ranges of frequencies in the signal. Furthermore, the power functions of the signal are replaced by corresponding convolution functions of the same order.

Efficient coding of high frequency signal information in a signal using a linear/non-linear prediction model based on a low pass baseband

In a digital audio broadcast system, to utilize transmission bandwidth efficiently, representations of a stereo audio signal for transmission are generated in accordance with an inventive multidescriptive coding technique. The representations, as generated, are then transmitted through multiple communication channels, respectively. The transmitted representations are received by a receiver where one or more of the representations are selected for recovery of the stereo audio signal. Because of the design of the multidescriptive coding used, the more representations are selected to recover the stereo audio signal, the higher the quality of the recovered signal.

Multidescriptive coding technique for multistream communication of signals

The invention provides methods and apparatus for processing information, e.g., audio, video or image information, for transmission in a communication system. In an illustrative embodiment, a joint multiple program coder determines the value of a single-bit or multiple-bit criticality measure, e.g., criticality flag, in a designated interval, e.g., the duration of an audio frame, for each of the programs in a set of multiple programs to be transmitted in the system. The joint multiple program coder allocates a pool of available bits to the programs based at least in part on the determined values of the criticality measures, such that a program with a higher-valued criticality measure in the designated time interval is allocated a greater percentage of the available bits for that interval than another one of the programs with a lower-valued criticality measure. The joint multiple program coder repeats the determination and allocation operations for each of a number of time intervals, e.g., frames, such that the bit allocation can vary from interval to interval.

Joint multiple program coding for digital audio broadcasting and other applications

An apparatus for providing at least first and second representations of an audio signal for use in a communications system is described. The apparatus comprises a first quantizer for quantizing at least a portion of the signal in accordance with a first multidimensional lattice to generate a first representation. The apparatus further comprises a second quantizer for quantizing at least a portion of the signal in accordance with a second, different multidimensional lattice to generate a second representation. In an illustrative embodiment, the first representation is a core representation containing core audio information. The second representation is an enhancement representation containing enhancement audio information. The core representation is necessary for recovering the audio signal with minimal acceptable quality. Audio quality is enhanced when the core representation, together with the enhancement representation, is used to recover the audio signal. A method for use in such an apparatus is also described.

Technique for multi-rate coding of a signal containing information

Audio tracks or other portions of a particular type of audio material to be encoded are analyzed to determine a value of at least one coding-related parameter suitable for providing optimal encoding of the particular type of audio material. When a given portion of the audio material is to be encoded for transmission in a perceptual audio coder of a communication system, the value of the coding-related parameter is identified and then utilized in conjunction with the encoding of the given portion. The determined value of the coding-related parameter may be at least a portion of a psychoacoustic model utilized in encoding the given portion of the particular type of audio material in the perceptual audio coder. As another example, the value of the coding-related parameter may be a setting of an audio processor utilized to process the given portion of the particular type of audio material prior to encoding the given portion in the perceptual audio coder.

Preclassification of audio material in digital audio compression applications

Improved error screening techniques are used in processing digital audio or other types of information received in a digital communication system. Control information associated with a given packet of the received information is identified and compared with a decoding requirement of the packet, in order to control the generation of an error indicator for the packet. More particularly, the error indicator may be generated in response to an inconsistency between the control information and the decoding requirement. For example, the control information may include an indication of packet length that can be compared to a number of bits required to decode the corresponding packet, with any inconsistency between the packet length indication and the number of required bits leading to the generation of an error flag for the packet. The invention is particularly well-suited for use in the processing of encoded digital audio information received in a hybrid in-band on-channel (HIBOC) digital audio broadcasting (DAB) system, but is also applicable to other types of information and other types of communication systems.

Error screening based on code and control information consistency in a communication system

Improved channel code configurations for use in transmission of digital audio or other types of information in a digital communication system. The channel code may include an outer channel code, e.g., a cyclic redundancy code (CRC), and an inner channel code, e.g., a complementary punctured pair convolutional (CPPC) code. In accordance with the invention, multiple code words of the outer code are associated with a given packet of the digital information, in accordance with a particular outer code configuration, so as to provide partial error flagging for different portions of the given packet. An information encoder, e.g., a PAC encoder, interacts with an outer code encoder to determine a bit allocation for transmission of packets at a particular bit rate, based at least in part on the outer code configuration. The invention is particularly well-suited for use in the transmission of digital audio information in a hybrid in-band on-channel (HIBOC) digital audio broadcasting (DAB) system, but is also applicable to other types of communication systems.

Channel code configurations for digital audio broadcasting systems and other types of communication systems

In a communications system, parametric coding in accordance with the invention is implemented to generate a representation of a stereo audio signal, which is composed of a left channel signal (L) and a right channel signal (R). To efficiently utilize transmission bandwidth, such a representation contains (1) information concerning only one of the L and R signals, and (2) parametric information based on which, together with (1), the other signal can be recovered. Because of the design of the parametric coding, the representation advantageously captures localization cues of the stereo audio signal, including intensity and phase characteristics of L and R. As a result, the stereo audio signal recovered from the transmitted representation affords a high stereo quality.

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Chatham, NJ, United States of America

Deepen Sinha