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: A system and method of retrieving a watermark in a watermarked signal are disclosed. The watermarked signal comprises odd and even overlapped blocks where the watermark is contained in the even blocks. The method comprises, for each k-th even block, subtracting the two adjacent odd numbered blocks from the k-th even block of the watermarked signal to retrieve s*k(n), transforming s*k(n) into the frequency domain to generate Sk(f), calculating a phase of Sk(f) as ?(f) and a phase of Sk(f) as ?(f), calculating the difference ?(f) between ?(f) and ?(f), unwrapping ?(f) to obtain the phase modulation {tilde over (?)}k(f), and using a Viterbi search to retrieve the watermark embedded in {tilde over (?)}k(f).

Abstract: A system, method and computer-readable medium are disclosed for using filters signal processing. The system includes a module that calculates a filter for each of a plurality of frequency bands, a module that groups the filters into a plurality of groups, a module that determines a representative filter for each group of the plurality of groups and a module that uses the representative filter of each group for frequency bands of the each group. The filters are temporal noise shaping filters (TNS) filters.

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: A method of retrieving a watermark in a watermarked signal involves processing a signal having odd- and even-overlapped blocks and where the watermark is contained in the even blocks. For each k-th block, the method comprises subtracting the odd-numbered blocks from the k-th block of the watermarked signal to generate s*k(n), applying an FFT to s*k(n) to generate a phase Sk(ƒ), calculating a phase of Sk(ƒ) as ?(ƒ) and a phase of an original signal Sk(ƒ) as ?(ƒ), calculating the difference ?(ƒ) between ?(ƒ) and ?(ƒ); and retrieving the watermark embedded in ?(ƒ). A Viterbi search may be the algorithm used to retrieve the watermark.

Abstract: A system, method and computer-readable medium are disclosed for using filter signal processing. The system conveys filter information for a spectrum for a signal at a receiver. The system comprises a module configured to transmit information regarding a first filter, a module configured to transmit information regarding a second filter and a module configured to transmit a mask to indicate switching between the first filter and the second filter across the spectrum. The filters may be temporal noise shaping (TNS) filters.

Abstract: A sound-capturing arrangement uses a set of directional microphones that lie approximately on a sphere having a diameter of 0.9 ms sound travel, which approximates the inter-aural time delay. Advantageously, one directional microphone points upward, one directional microphone points downward, and the odd number of microphones are arranged relatively evenly in the horizontal plane. On one embodiment, the signals from the microphones that point upward and downward are combined with the signals of the horizontal microphones before the signals of the horizontal microphones are transmitted or recorded.

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: A system and method of retrieving a watermark in a watermarked signal are disclosed. The watermarked signal comprises odd and even overlapped blocks where the watermark is contained in the even blocks. The method comprises, for each k-th even block, subtracting the two adjacent odd blocks from the k-th even block of the watermarked signal to retrieve {overscore (s)}*k(n), transforming {overscore (s)}*k(n) into the frequency domain to generate {overscore (S)}k(f), calculating a phase of {overscore (S)}k(f) as {overscore (?)}(f) and a phase of Sk(f) as ?(f), calculating the difference ?(f) between {overscore (?)}(f) and ?(f), unwrapping ?(f) to obtain the phase modulation {tilde over (?)}k(f), and using a Viterbi search to retrieve the watermark embedded in {tilde over (?)}k(f).

Abstract: In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The 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. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions 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.

Abstract: Better reproduction of sound is achieved analyzing audio signals for perceptual soundfield imaging cues, and applying appropriate portions of the audio signal to a directional speaker and to a diffusing speaker, based on the analyzed audio signals. Specifically, the audio signal is applied to a critical band filter bank, and at the higher frequencies a detector detects leading edges in the signal's envelope. Concurrently, the audio signal is applied to a delay element followed by a reconstruction filter bank. The output of each filter in the reconstruction filter bank is applied to a one-input/two-outputs soft switch (single pole, double throw) that is controlled by one or more of the detectors. One output of the switch is applied to a directional speaker, and the other output of the switch is applied to a diffusing speaker.

Abstract: A sound-capturing arrangement uses a set of directional microphones that lie approximately on a sphere having a diameter of 0.9 ms sound travel, which approximates the inter-aural time delay. Advantageously, one directional microphone points upward, one directional microphone points downward, and the odd number of microphones are arranged relatively evenly in the horizontal plane. On one embodiment, the signals from the microphones that point upward and downward are combined with the signals of the horizontal microphones before the signals of the horizontal microphones are transmitted or recorded.

Abstract: A sound-capturing arrangement uses a set of directional microphones that lie approximately on a sphere having a diameter of 0.9 ms sound travel, which approximates the inter-aural time delay. Advantageously, one directional microphone points upward, one directional microphone points downward, and the odd number of microphones are arranged relatively evenly in the horizontal plane.

Abstract: The present invention provides a system and method for storing re-synchronization, error correction and/or error detection data within an existing communication protocol, while still maintaining full compliance to a standard, such as the MPEG-2 AAC standard. By doing so, data information can still be passed from an encoder to a decoder via a channel using an existing and well known standard transport protocol. However, the existing well known transport protocol can now include the data necessary for synchronization of the decoder to the received raw data, along with error detection and error correction by the decoder.

Abstract: In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The 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. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions 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.

Abstract: The present invention provides a system and method for storing re-synchronization, error correction and/or error detection data within an existing communication protocol, while still maintaining full compliance to a standard, such as the MPEG-2 AAC standard. By doing so, data information can still be passed from an encoder to a decoder via a channel using an existing and well known standard transport protocol. However, the existing well known transport protocol can now include the data necessary for synchronization of the decoder to the received raw data, along with error detection and error correction by the decoder.

Abstract: Perceptual coding is accomplished by measuring the envelope roughness of the filtered audio signal, which may be directly converted to the noise to mask threshold needed to calculate the perceptual threshold or “just noticeable difference”. Thus, the present invention does not require any complex calculations to determine tonality, either by a measure of predictability or by the calculation of a loudness or loudness uncertainty. Instead, the envelope roughness of the signal is simply reduced directly to the noise to mask ratio.

Abstract: A method and apparatus for quantizing audio signals is disclosed which advantageously produces a quantized audio signal which can be encoded within an acceptable range. Advantageously, the quantizer uses a scale factor which is interpolated between a threshold based on the calculated threshold of hearing at a given frequency and the absolute threshold of hearing at the same frequency.

Abstract: A method and apparatus for quantizing audio signals is disclosed which advantageously produces a quantized audio signal which can be encoded within an acceptable range. Advantageously, the quantizer uses a scale factor which is interpolated between a threshold based on the calculated threshold of hearing at a given frequency and the absolute threshold of hearing at the same frequency.

Abstract: A method is disclosed for determining estimates of the perceived noise masking level of audio signals as a function of frequency. By developing a randomness metric related to the euclidian distance between (i) actual frequency components amplitude and phase for each block of sampled values of the signal and (ii) predicted values for these components based on values in prior blocks, it is possible to form a tonality index which provides more detailed information useful in forming the noise masking function. Application of these techniques is illustrated in a coding and decoding context for audio recording or transmission. The noise spectrum is shaped based on a noise threshold and a tonality measure for each critical frequency-band (bark).