Abstract: A method of identifying patterns in a digitized acoustic signal is disclosed. The method comprises: (i) converting the digitized acoustic signal into a spatial representation being defined by a plurality of regions on a vibrating membrane, each the regions having a different vibration resonance, each the vibration resonance corresponding to a different frequency of the acoustic signal; (ii) iteratively calculating a weight function, the weight function having a spatial dependence being representative of acoustic patterns of each region of the plurality of regions; and (iii) using the weight function for converting the spatial representation into a reconstructed acoustic signal; thereby identifying the patterns in the acoustic signal.

Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. At least two transforms are applied on an audio signal, each with different transform periods for better resolutions at both low and high frequencies. The transform coefficients are selected and combined such that the data rate remains similar as a single transform. The transform coefficients may be coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: Various embodiments of the present invention are directed to a frequency-domain coder/decoder for an audio-conference communication system that includes acoustic-echo-cancellation functionality. In one embodiment of the present invention, an acoustic echo canceller is integrated into the frequency-domain coder/decoder and ameliorates or removes acoustic echoes from audio signals that have been transformed to the frequency domain and divided into subbands by the frequency-domain coder/decoder.

Abstract: Provided are a method, medium and apparatus for enhancing an acoustic signal using an auditory property. An acoustic signal is enhanced by generating a plurality of harmonic signals based on a predetermined acoustic signal frequency, selecting harmonic signals, which exist in an area masked by the predetermined harmonic signal, from among the generated plurality of harmonic signals, and outputting harmonic signals remaining after excluding the selected harmonic signals from the generated plurality of harmonic signals. The enhancement results in a bass signal of good sound quality and having a low distortion ratio, without changing the structure of a micro speaker.

Abstract: Estimating a signal power in a compressed audio signal [A] is provided, the audio signal comprising blocks of quantized samples, a given block being provided with a set of scale factors. The estimating is performed by extracting the set of scale factors from the compressed audio signal, and estimating the signal power in the given block based on a combination of the scale factors. Advantageously, the extracting step and estimating step are performed on only a sub-set of the set of scale factors. The signal power estimation may be used in a silence detector (11) for use in a receiver (1).

Abstract: The invention concerns a frequency-domain error concealment technique for information that is represented, on a frame-by-frame basis, by coding coefficients. The basic idea is to conceal an erroneous coding coefficient by exploiting coding coefficient correlation in both time and frequency. The technique is applicable to any information, such as audio, video and image data, that is compressed into coding coefficients and transmitted under adverse channel conditions. The error concealment technique proposed by the invention has the clear advantage of exploiting the redundancy of the original information signal in time as well as frequency. For example, this offers the possibility to exploit redundancy between frames (inter-frame) as well as within frames (intra-frame). The use of coding coefficients from the same frame as the erroneous coding coefficient is sometimes referred to as intra-frame coefficient correlation and it is a special case of the more general frequency correlation.

Abstract: An audio data stream from a processing system may be buffered to allow low power states in the processing system during audio playback. An audio buffer may be provided external to the processing system and between the processing system and an audio codec. The audio buffer may also shift to an alternate audio data interface mode when the processing system is in the low power state. Of course, many alternatives, variations, and modifications are possible without departing from this embodiment.

Abstract: Many compressed audio or video frames contain silence (if audio), or a blank image (if video); these essentially information content free (e.g. silent if audio or blank if video) frames can be both detected whilst still in compressed form and then used to carry the additional data. In an MPEG implementation, subbands associated with silent frames are rendered digitally silent and then used to carry PAD (Programme Associated Data).

Abstract: Prior to embedding a watermark in an audio signal, a spectral representation of the audio signal and a spectral representation of the watermark signal are determined. The spectral representation of the watermark signal is then processed on the basis of a psychoacoustic masking threshold of the audio signal. The processed watermark signal is combined with the audio signal to obtain an audio signal bearing a watermark.

Abstract: An apparatus for scalable encoding a spectrum of a signal including audio and/or video information, with the spectrum comprising binary spectral values, includes a means for generating a first sub-scaling layer and a second sub-scaling layer in addition to a means for forming the encoded signal, with the means for forming being implemented so as to include the first sub-scaling layer and the second sub-scaling layer into the encoded signal that the first and the second sub-scaling layer are separately decodable from each other. In contrast to a full-scaling layer, a sub-scaling layer includes only the bits of a certain order of a part of the binary spectral values in the band, so that, by additionally decoding a sub-scaling layer, a more finely controllable and a more finely scalable precision gain may be achieved.

Abstract: For taking account of different requirements on encoded audio data having different bit rates, at least two different amounts of pre-processing are applied to an audio signal to obtain at least two different target signals. A first one of the target signals is then encoded to obtain primary coded data. At least a second one of the at least two different target signals is moreover used for generating enhancement data for the primary coded data.

Abstract: An apparatus for processing a multi-channel signal includes a means for determining a similarity between a first one of two channels and a second one of the two channels. Furthermore, a means for performing a prediction filtering of the spectral coefficients is provided, which is formed to perform a prediction filtering with only a single prediction filter for both channels in case of high similarity between the first and the second channel, and to perform a prediction filtering with two separate prediction filters in case of a dissimilarity between the first and the second channel. With this, an introduction of stereo artifacts and a deterioration of the coding gain in stereo coding techniques are avoided.

Abstract: The present invention provides an apparatus, method and tangible medium storing instructions for determining tonality of an input audio signal, for selection of corresponding masked thresholds for use in perceptual audio coding. In the various embodiments, the input audio signal is sampled and transformed using a compressed spectral operation to form a compressed spectral representation, such as a cepstral representation. A peak magnitude and an average magnitude of the compressed spectral representation are determined. Depending upon the ratio of peak-to-average magnitudes, a masked threshold is selected having a corresponding degree of tonality, and is used to determine a plurality of quantization levels and a plurality of bit allocations to perceptually encode the input audio signal with a distortion spectrum beneath a level of just noticeable distortion (JND).

Abstract: Methods and apparatus are provided for the creation and utilization of unique compressed data stream compositions, structures and formats which allow for the alteration of the data stream's data rate without first decoding the data stream back to its uncompressed form and then re-encoding the resulting uncompressed data at a different data rate. Such methods and apparatus perform this data rate alteration, known as scaling, such that optimal quality is maintained at each scaled data rate, while performing said scaling with low computational complexity. In addition, the present invention provides for data rate alteration in small increments. A unique application for the disclosed bit rate scaling method and apparatus is also described.

Abstract: Methods and apparatus are provided for communicating an audio stream. A perceptual mask is estimated for an audio stream, based on the perceptual threshold of the human auditory system. A hidden sub-channel is dynamically allocated substantially below the estimated perceptual mask based on the characteristics of the audio stream, in which additional payload is transmitted. The additional payload can be related to components of the audio stream that would not otherwise be transmitted in a narrowband signal, or to concurrent services that can be accessed while the audio stream is being transmitted. A suitable receiver can recover the additional payload, whereas the audio stream will be virtually unaffected from a human auditory standpoint when received by a traditional receiver. A coding scheme is also provided in which a portion of a codec is used to code an upper-band portion of an audio stream, while the narrowband portion is left uncoded.

Abstract: Methods and devices for dynamically adjusting a multi-band signal-modification profile based on a psychoacoustic model are disclosed. In one arrangement, the encoding parameter side information is used to estimate encoding noise of an encoded signal. The signal spectrum of the signal is estimated. Adjustments to the multi-band signal-modification profile are determined using the estimated noise and signal spectrum and a psychoacoustic profile.

Abstract: Techniques for determining scale factor values when encoding audio data are described. According to one technique, a particular scale factor value (SFV) is estimated using an audio quality estimator function that is non-linear. After a certain point, a decrease in noise results in a smaller increase in audio quality. According to another technique, an initial SFV is estimated for each scale factor band (SFB). When estimating the cost of transitioning from one SFB to another, only a proper subset of possible SFVs are considered. The proper subset is based, at least in part, on the initial SFV.

Abstract: A method and system processes a speech signal. A fast Fourier transform is performed on a speech signal to produce a speech signal having a plurality of frequency bands in a frequency domain.

Abstract: In an audio coding system, an encoding transmitter represents encoded spectral components as normalized floating-point numbers. The transmitter provides first and second control parameters that may be used to transcode the encoded spectral parameters. A transcoder uses first control parameters to partially decode the encoded components and uses second control parameters to re-encode the components. The transmitter determines the second control parameters by analyzing the effects of arithmetic operations in the partial-decoding process to identify situations where the floating-point representations lose normalization. Exponents associated with the numbers that lose normalization are modified and the modified exponents are used to calculate the second control parameters.

Abstract: Objective measurement methods and devices for predicting perceptual quality of speech signals degraded in speech processing/transporting systems have unreliable prediction results in cases where the degraded and reference signals show in between severe timbre differences. Improvement is achieved by applying a partial compensation step within in a signal processing stage using a frequency dependently clipped compensation factor for compensating power differences between the degraded and reference signals in the frequency domain. Preferably clipping values for clipping the compensation factor have larger frequency-dependency in a range of low frequencies with respect to a centre frequency of the human auditory system, than in a range of high frequencies.

Abstract: An adaptive variable bit rate audio encoder and method that examines audio level information and detects various information in the audio data in a psychoacoustic model to create a quantization value and assign a mode tag to a single frame of audio. A bit rate is assigned according to one of three modes, a self-adaptive mode which is free-running and takes direction only from the characteristics of the incoming audio signal, a managed mode which is controlled by rules set from a statistical multiplexer, and a combination of self-adaptive and managed in which control rules from the statistical multiplexer act to maintain limits on the self-adaptive mode.

Abstract: A method for objective speech quality assessment that accounts for phonetic contents, speaking styles or individual speaker differences by distorting speech signals under speech quality assessment. By using a distorted version of a speech signal, it is possible to compensate for different phonetic contents, different individual speakers and different speaking styles when assessing speech quality. The amount of degradation in the objective speech quality assessment by distorting the speech signal is maintained similarly for different speech signals, especially when the amount of distortion of the distorted version of speech signal is severe. Objective speech quality assessment for the distorted speech signal and the original undistorted speech signal are compared to obtain a speech quality assessment compensated for utterance dependent articulation.

Abstract: Disclosed is an objective speech quality assessment technique that reflects the impact of distortions which can dominate overall speech quality assessment by modeling the impact of such distortions on subjective speech quality assessment, thereby, accounting for language effects in objective speech quality assessment.

Abstract: Systems and methods for audio signal processing are provided. In exemplary embodiments, a filter cascade of complex-valued filters are used to decompose an input audio signal into a plurality of frequency components or sub-band signals. These sub-band signals may be processed for phase alignment, amplitude compensation, and time delay prior to summation of real portions of the sub-band signals to generate a reconstructed audio signal.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding.

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: The indexing method comprises forming a set of tracks of pulse positions, restraining the positions of the non-zero-amplitude pulses of the combinations of the codebook in accordance with the set of tracks of pulse positions, and indexing in the codebook each non-zero-amplitude pulse of the combinations at least in relation to the position of the in the corresponding track, the amplitude of the pulse, and the number of pulse positions in said corresponding track. For indexing the position(s) of one and two non-zero amplitude pulse(s) in one track, procedures code—1 pulse and code—2 pulse are respectively used. When the positions of a number X of non-zero-amplitude pulses are located in one track, X?3, subindices of these X pulses are calculated using the procedures code—1 pulse and code—2 pulse, and a global index is calculated by combining these subindices.

Abstract: When encoding an audio signal, the audio signal is first encoded with the first encoder to obtain a first encoder output signal. This first encoder output signal is written into a bit stream. It is further decoded by a decoder to provide a decoded audio signal. The decoded audio signal is compared with the original audio signal to obtain a residual signal. The residual signal is then encoded via a second encoder to provide a second encoder output signal which is also written into a bit stream. The first encoder has a first time or frequency resolution. The second encoder has a second time or frequency resolution. The first resolution differs from the second resolution, so that in a respective decoder, an audio signal with both a high time resolution as well as a high frequency resolution can be retrieved.

Abstract: A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.

Abstract: Method and apparatus for environment detection and adaptation in hearing assistance devices. Performance of feature extraction and environment detection to perform adaptation to hearing assistance device operation for a number of hearing assistance environments. The system detecting various noise sources independent of speech. The system determining adaptive actions to take place based on predicted sound class. The system providing individually customizable response to inputs from different sound classes. In various embodiments, the system employing a Bayesian classifier to perform sound classifications using a priori probability data and training data for predetermined sound classes. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.

Abstract: According to one aspect of the invention, a method is provided in which audio samples representing an input audio signal are received. The input audio samples are transformed into a vector of spectral values in a frequency domain. A value of a quantizing parameter is determined that satisfies one or more criteria based, at least in part, on a modified Newtonian search process, the determined value of the quantizing parameter being used to quantize the respective vector of spectral values to generate a vector of quantized values.

Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.

Abstract: An audio information retrieval method, medium, and system that can rapidly retrieve audio information, even in noisy environments, by extracting a modulation spectrum that is robust against noise, converting features of the extracted modulation spectrum into hash bits, and using a hash table. The audio information retrieval method may include extracting a modulation spectrum from audio data of a compressed domain, converting the extracted modulation spectrum into fingerprint bits, arranging the fingerprint bits in a form of a hash table, converting a received query into an address by a hash function corresponding to the query, and retrieving the audio information by referring to the hash table.

Abstract: Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

Abstract: An improved noise canceling microphone is provided including robust design features and advanced noise control and speech discrimination convergence characteristics. Two adaptive controllers are used to ensure robust performance in quickly changing acoustic environments ensuring an acceptable minimum performance characteristic. Additionally, a new real-time spectral estimation procedure is applied to a noise canceling communications microphone platform that permits continued and optimal adaptation of non-voice bandwidth frequencies during speech transients.

Abstract: A method for stereo audio perceptual encoding of an input signal includes masking threshold estimation and bit allocation. The masking threshold estimation and bit allocation are performed once every two encoding processes. Another method for stereo audio perceptual encoding of an input signal includes performing a time-to-frequency transformation, performing a quantization, performing a bitstream formatting to produce an output stream, and performing a psychoacoustics analysis. The psychoacoustics analysis includes masking threshold estimation on a first of every two successive frames of the input signal.

Abstract: An audio compression method using wavelet packet transform (WPT) in MPEG1 layer 3 (hereinafter referred to as “MP3”) and a system thereof are provided. The method comprises calculating perceptual energy by analyzing audio samples which are input based on a psychoacoustic model; according to comparison of the level of the calculated perceptual energy with a threshold, selectively determining a modified DCT (MDCT) processing window and a wavelet packet transform (WPT) processing window; by processing audio samples corresponding to the scopes of the determined windows in the MDCT and WPT, converting the audio samples into data on frequency domains; and quantizing the processed data on the frequency domains according to the number of assigned bits.

Abstract: A system for transmitting audio signals over a telecommunications link generates the signals as two or more alternative feeds, for example at different data rates. The two feeds are encoded using coding methods having a frame structure with different frame lengths. To facilitate switching between the two, the input signal is notionally divided into temporal portions and each is coded by taking it, plus enough of the next (or preceding) portion to make up a whole number of frames, and encoding it, whereby the encoded portions overlap—at least for one of the feeds. The overlap is lost upon decoding by discarding duplicate material.

Abstract: A sound processor including a microphone (1), a pre-amplifier (2), a bank of N parallel filters (3), means for detecting short-duration transitions in the envelope signal of each filter channel, and means for applying gain to the outputs of these filter channels in which the gain is related to a function of the second-order derivative of the slow-varying envelope signal in each filter channel, to assist in perception of low-intensity short-duration speech features in said signal.

Abstract: The quality of audiovisual material is assessed by measuring the audio an video quality and computing from these a combined measure. Using a parameter indicative of the degree of motion represented by the video, the computation employs one of a plurality of algorithms selected in dependence on the value of the parameter.

Abstract: Described herein is a technology for facilitating the recognition and categorization of the content of digital signals. This abstract itself is not intended to limit the scope of this patent. The scope of the present invention is pointed out in the appending claims.

Abstract: A device and a method for compressing signal information by removing (thinning out) the signal component of a signal in a specific frequency band. Firstly, an input time-series signal (e.g., a PCM signal) is converted by an analyzer (11) into a spectrum signal. Next, of the bands obtained by dividing the spectrum equally into bands, the band having a predetermined or higher correlation in the spectrum distribution with the lower frequency band is specified as a harmonic band by a frequency band masking unit (12). Then, a removal band from which the spectrum is to be removed is determined from the harmonic band, and the spectrum signal of this removal band, from which the spectrum component has been removed (namely the frequency component has been thinned out), is fed to a synthesizer (13).

Abstract: A method and apparatus for audio compression receives an audio signal. Transform coding is applied to the audio signal to generate a sequence of transform frequency coefficients. The sequence of transform frequency coefficients is partitioned into a plurality of non-uniform width frequency ranges and then zero value frequency coefficients are inserted at the boundaries of the non-uniform width frequency ranges. As a result, certain of the transform frequency coefficients that represent high frequencies are dropped.

Abstract: Auditory-articulatory analysis for use in speech quality assessment. Articulatory analysis is based on a comparison between powers associated with articulation and non-articulation frequency ranges of a speech signal. Neither source speech nor an estimate of the source speech is utilized in articulatory analysis. Articulatory analysis comprises the steps of comparing articulation power and non-articulation power of a speech signal, and assessing speech quality based on the comparison, wherein articulation and non-articulation powers are powers associated with articulation and non-articulation frequency ranges of the speech signal.

Abstract: A process and system for providing objective quality measurement of a target audio signal. Reference and target signals are processed by a peripheral ear processor, and compared to provide a basilar degradation signal. A cognitive processor employing a neural network then determines an objective quality measure from the basilar degradation signal by calculating certain key cognitive model components.

Abstract: The present invention is generally directed to a system and method for enhancing speech using a multi-channel noise filtering process that is based on psychoacoustic masking effects. A speech enhancement/noise reduction scheme according to the present invention is designed to satisfy the psychoacoustic masking principle and to minimize the signal total distortion by exploiting multiple microphone signals to enhance the useful speech signal at reduced level of artifacts.

Abstract: An audio processing tool measures the quality of reconstructed audio data. For example, an audio encoder measures the quality of a block of reconstructed frequency coefficient data in a quantization loop. The invention includes several techniques and tools, which can be used in combination or separately. First, before measuring quality, the tool normalizes the block to account for variation in block sizes. Second, for the quality measurement, the tool processes the reconstructed data by critical bands, which can differ from the quantization bands used to compress the data. Third, the tool accounts for the masking effect of the reconstructed data, not just the masking effect of the original data. Fourth, the tool band weights the quality measurement, which can be used to account for noise substitution or band truncation. Finally, the tool changes quality measurement techniques depending on the channel coding mode.

Abstract: A data-compressed audio waveform is temporally modified without requiring complete decompression of the audio signal. Packets of compressed audio data are first unpacked, to remove scaling that was applied in the formation of the packets. The unpacked data is then temporally modified, using one of a number of different approaches. This modification takes place while the audio information remains in a data-compressed format. New packets are then assembled from the modified data, to produce a data-compressed output stream that can be subsequently processed in a conventional manner to reproduce the desired sound. The assembly of the new packets employs a technique for inferring an auditory model from the original packets, to requantize the data in the output packets.

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.