Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A system and method of generating a watermarked signal are disclosed. The system segments the signal into overlapping blocks using a window function and processes the overlapping blocks according to whether each block is odd- or even-numbered. The system windows the odd-numbered blocks, modulates the phase of each block in the frequency domain, transforms each modulated block in the time domain, windows each block transformed into the time domain and overlap-adds each odd-numbered block with each even-numbered block to generate the watermarked signal.

Abstract: System, method and computer-readable medium are disclosed for using filters signal processing. The system includes a module that receives information regarding a first filter, a module that receives information regarding a second filter, and a module that receives date to indicate switching between the first filter and the second filter across the spectrum of the received audio signal, and a module that processes the received audio signal according to the received data and switching between the first filter and the second filter, wherein at least one of the first filter and the second filter represent a merger of two initial filters.

Abstract: A system, method and computer readable medium that processing a watermarked signal using the phase Sk(f) of an original signal. The watermarked signal includes odd and even overlapped blocks where the watermark is contained in the even blocks. The method comprises test-decoding the watermarked signal and, if the watermarked signal contains errors, recoding the watermarked signal with a higher redundancy code. The steps of test-decoding and recording may be performed until all errors are corrected.