Abstract: 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.

Abstract: Datacasting bandwidth enhancements are accomplished using a perceptual audio coder (PAC) that indicates, to a push-pull gateway (PPG), the availability of deterministic and opportunistic bandwidth. In case of real-time programs, data is pulled from a pre-managed datacasting service queue, and the deterministic bandwidth is used for real-time low bit-rate program related information and the opportunistic bandwidth is used for transmission of encoded non-real-time data content. For recorded content, further enhancement is achieved by preprocessing the audio song. The PAC identifies data rate (e.g., in kbps) and the deterministic and opportunistic deterministic bandwidth available (with delay variance). This information is conveyed to a push-pull gateway which performs management of data content destined for the deterministic opportunistic queue.

Abstract: 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.

Abstract: A dual-mode receiver that functions both as a digital audio broadcast receiver and a solid-state digital audio content player is described. In a “radio” mode, the receiver is able to: receive a digital audio bitstream, decode the received bitstream, store or render the decoded bitstream either based on an input from a user or a pre-programmed instruction from a user. In a “play” mode, the receiver is able to access the stored digital audio content, or extract stored digital audio content from a compact media device (e.g., compact flash memory) that is interfaced with the receiver.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: In a communications system implementing, e.g., an in-band on channel AM (IBOC-AM) (also known as “hybrid IBOC-AM”) scheme, multiple bit streams are used to represent an audio signal to be transmitted over one or more frequency bands including, e.g., parts of an AM frequency band for radio broadcast. These bit streams contain various and/or equivalent amounts of audio information. In an illustrative embodiment, at least one of the bit streams is a core bit stream containing core audio information. The remaining bit streams are enhancement bit streams containing enhancement audio information. The core bit stream is necessary for recovering the audio signal with minimal acceptable quality. Such quality is enhanced when the core bit stream, together with one or more of the enhancement bit streams, is used to recover the audio signal. In accordance with the invention, the AM frequency band is divided into subbands.

Abstract: In a communications system, multi-rate coding in accordance with the invention is implemented to generate multiple representations of an audio signal at different rates. These representations contain equivalent and/or various amounts of audio information. In an illustrative embodiment, at least one of the representations is a core representation containing core audio information. The remaining representations are enhancement representations containing enhancement audio information. The core representation is necessary for recovering the audio signal with minimal acceptable quality. Such quality is enhanced when the core representation, together with one or more of the enhancement representations, is used to recover the audio signal.

Abstract: An audio information bit stream including audio control bits and audio data bits is processed for transmission in a communication system. The audio data bits are first separated into n classes based on error sensitivity, that is, the impact of errors in particular audio data bits on perceived quality of an audio signal reconstructed from the transmission. Each of the n different classes of audio data bits is then provided with a corresponding one of n different levels of error protection, where n is greater than or equal to two. The invention thereby matches error protection for the audio data bits to source and channel error sensitivity. The audio control bits may be transmitted independently of the audio data bits, using an additional level of error protection higher than that used for any of the n classes of the audio data bits. Alternatively, the control bits may be combined with one of the n classes of audio data bits and provided with the highest of the n levels of error protection.

Abstract: Methods and apparatus for decoding multiple program information, e.g., audio, video or image information, in a communication system are disclosed. In an illustrative embodiment, a multiple program decoder includes a deinterleaver for deinterleaving information corresponding to a set of frames, using a specified deinterleaving length. A given one of the frames includes information from each of at least a subset of the programs, and the frames are encoded using an outer code, e.g., a CRC code, RS code, BCH code or other type of linear block code, and an inner code, e.g., a convolutional code, turbo code or trellis coded modulation. The multiple program decoder includes an inner code decoder for decoding the inner code over one or more of the programs, and an outer code decoder for decoding the outer code for a selected one of the programs.

Abstract: In a communications system, multi-rate coding in accordance with the invention is implemented to generate multiple representations of an audio signal at different rates. These representations contain equivalent and/or various amounts of audio information. In an illustrative embodiment, at least one of the representations is a core representation containing core audio information. The remaining representations are enhancement representations containing enhancement audio information. The core representation is necessary for recovering the audio signal with minimal acceptable quality. Such quality is enhanced when the core representation, together with one or more of the enhancement representations, is used to recover the audio signal.

Abstract: Methods and apparatus for processing information, e.g., audio, video or image information, for transmission in a communication system are disclosed. In an illustrative embodiment, a multiple program coder encodes a bitstream for each of the programs, in accordance with a specified bit allocation. An outer code encoder receives the encoded bitstreams from the multiple program coder, and jointly applies an outer code, e.g., a CRC code, RS code, BCH code or other type of linear block code, to subblocks of a subset of the resulting encoded bitstreams. The outer code is applied such that subblocks of each of the encoded bitstreams in the subset of the encoded bitstrcams are associated with a single outer code block. The number of subblocks associated with a given outer code block may be an integer or non-integer number of code blocks.

Abstract: Methods and apparatus for processing information, e.g., audio, speech, video or image information, for transmission in a communication system. In an illustrative embodiment, a set of bit streams are generated from an audio information signal. The set of bit streams may be, e.g., a total of four bit streams generated by separating each of two multiple description bit streams, corresponding to separate representations of the audio information signal, into first and second class bit streams. The first and second class bit streams associated with the first multiple description bit stream may then be transmitted in respective first and second subbands of a first sideband of an FM host carrier, while the first and second class bit streams associated with the second multiple description bit stream are transmitted in respective first and second subbands of a second sideband of the FM host carrier.

Abstract: 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, interference characteristics are determined for a set of n channels to be used to transmit audio information bits, where n is greater than or equal to two. The audio information bits are separated into n classes based on error sensitivity, for example, the impact of errors in particular audio data bits on perceived quality of an audio signal reconstructed from the transmission. The classes of bits are then assigned to the n channels such that the classes of bits having the greatest error sensitivity are transmitted over the channels which are the least susceptible to interference.

Abstract: 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, the value of a single-bit or multiple-bit criticality flag is determined for each of the programs in a set of multiple programs to be transmitted in the system. The information bits for each of the programs are then separated into n classes, where n is greater than or equal to two, based on the values of the criticality flags for the programs. Each of the classes is provided with a different level of error protection, e.g., through the use of different convolutional codes or other suitable techniques. The program or programs having the highest criticality flag values in a given frame or other designated time interval thus have a larger percentage of their information bits assigned to the class that is provided with the highest level of error protection.

Abstract: An audio coding technique which utilizes a signal adaptive switched filterbank having a first filterbank and a wavelet filterbank. The filterbank switches between the first filterbank and the wavelet filterbank to filter an input signal as a function of the stationarity of the input signal. The first filterbank is utilized to filter stationary signal components. The wavelet filterbank is utilized to filter non-stationary signal components (e.g., attacks).

Abstract: A five-channel perceptual audio compression system which encodes five matrixed channels switches among a fourteen encoding modes each utilizing a respective different set of matrixed channels. Six of the modes are for encoding the three front channels and eight of the modes are for encoding the two back channels. The rate of perceptual encoding the matrixed channels is controlled by adjusting individual noise thresholds as a function of a global masking threshold.

Abstract: A new technique for the determination of the masking effect of an audio signal is employed to provide transparent compression of an audio signal at greatly reduced bit rates. The new technique employs the results of recent research into the psycho-physics of noise masking in the human auditory system. This research suggests that noise masking is a function of the uncertainty in loudness as perceived by the brain. Measures of loudness uncertainty are employed to determine the degree to which audio signals are "tone-like" (or "noise-like"). The degree of tone-likeness, referred to as "tonality," is used to determine masking thresholds for use in the compression of audio signals. Tonality, computed in accordance with the present invention, is used in conventional and new arrangements to achieve compression of audio signals.

Abstract: A five-channel perceptual audio compression system which encodes five matrixed channels switches among a fourteen encoding modes each utilizing a respective different set of matrixed channels. Six of the modes are for encoding the three front channels and eight of the modes are for encoding the two back channels. A number of the modes have as at least one of their respective matrixed channels a matrixed channel which is a function of a) one of the input channels or a sum or difference of two of them, and b) a predicted value of a).