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: 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: An independent transport layer is used as an error correction mechanism in a method employing perceptual audio coding. The transport layer is characterized by the addition of information, via a transport header, to each of a succession of blocks of encoded program material which is independent of the information within its respective block of encoded program material. The transport header comprises information on the structure of the block. Thus, the transport header is easily separated from the digital representation of the program material and is independent of the content of the digital representation thereby providing for changing of the content of the digital representation without changing the transport layer.

Abstract: An audio coding technique is presented, which utilizes a time varying non-uniform filterbank in conjunction with a Perceptual Audio Coder. The non-uniform filterbank is formed from a plurality of uniform filter bank sections, and a transition filter. Each uniform filterbank section covers a portion of a predetermined frequency axis, and the transition filter is utilized to cover the remainder of the predetermined frequency axis. Aliasing terms introduced at the band edges of the transition filter are cancelled to obtain an overall transfer function for the non-uniform filterbank, which has linear phase and is approximately flat. Use of the non-uniform filterbank permits an increase in temporal resolution, which results in an improved coding technique over a wide range of frequencies.

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 method of encoding an audio signal is disclosed. The method comprises partitioning the audio signal into a first time block and a second time block. Next, a first time block first energy value and a first time block second energy value are calculated. Next, a second time block first energy value and a second time block second energy value are calculated. Next, the technique determines if an attack has occurred in the second time block by comparing the second time block first energy value and the second time block second energy value and also comparing the Erst time block and the second time block Advantageously, the method identifies attach such that the decoder can reproduce the attacks with little audible distortion and also affords the advantage of using long windows for portions of the audio signal that do not contain attacks.

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 form noise masking thresholds for use in the compression of audio signals. These measures are employed in an illustrative subband, analysis-by-synthesis framework. In accordance with the illustrative embodiment, provisional encodings of the audio signal are performed to determine the encoding which achieves a loudness differential, between the original and coded audio signal, which is less than (but not too far below) the loudness uncertainty.

Abstract: An image-processing system for perceptual coding of an image is disclosed. Coding is accomplished through an analysis of human visual sensitivity to noise in halftone images and an analysis of one or more signals representing the image to be coded. These analyses determine levels of noise. A first image is encoded so as to produce encoded values without introducing noise which exceeds a determined level of noise. Analysis of human visual sensitivity is carried out under a set of user determined conditions comprising viewing distance and lighting. The encoded image is communicated and decoded to produce representation of the first image for display. As part of displaying the representation of the first image, a halftoning process is carried out. The halftoning process may comprise the interpolation of data values of the representation of the first image as well as the addition of micro-dither to the representation.

Abstract: A controllable-gain amplifier is connected between the transmitter and receiver of a telephone set and is controlled in response to the level of the telephone user's speech so as to provide an increase in gain when the user speaks too loud and a decrease in gain when the user speaks too low. This results in a sidetone which causes the user to change his or her voice level to obtain an acceptable level of sidetone. A better-controlled user voice level is thus achieved.

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).