Abstract: A system for enhancement of noisy speech comprises an input unit is configured to subdivide the spectrum of the input signal into a plurality of frequency sub-bands and to provide time-frequency coefficients X(k,m) for a sequence [X(k,m??D+1) . . . X(k,m?)] of observable noisy signal samples for each of said frequency sub-bands, where k and m are frequency and time indices, respectively, and D is larger than 1. The system further comprises enhancement processing unit configured to receive X(k,m) and to provide enhanced time-frequency coefficients ?(k,m), a storage for statistical model(s) of speech and for statistical model(s) of noise, and an optimizing unit configured to provide said enhanced time-frequency coefficients ?(k,m) using said statistical model of speech and said statistical model of noise, while considering said sequence [X(k,m??D+1) . . . X(k,m?)] of observable noisy signal samples.

Abstract: An audio processing circuit, audio unit and integrated circuit and method for blending are described. The audio processing circuit comprises: a time-to-frequency domain circuit configured to convert a primary audio signal and a secondary audio signal in a time-domain form and output frequency spectrum representations thereof; a cross-fade magnitude circuit configured to obtain a cross-faded magnitude spectrum representation from the frequency spectrum representations; a cross-fade phase circuit configured to obtain a separate cross-faded phase spectrum representation from the frequency spectrum representations; and a frequency-to-time domain circuit configured to receive the separate cross-faded magnitude and cross-faded phase spectrum representations and output a time domain signal.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed to reduce noise from harmonic noise sources. Example methods disclosed herein include determining a first point representing a comparatively large amplitude for a frequency value in an audio sample. Disclosed example methods also include generating a first contour trace of points having amplitude, frequency and phase values within thresholds from the first point of comparatively large amplitude. Example methods include generating a second contour trace of points having amplitude, frequency and phase values within thresholds of a second point of comparatively large amplitude. Disclosed example methods include calculating a parameter for each of the contour traces and determining if the first or second contour traces represent outliers based on the parameters. Disclosed example methods also include, in response to determining the contour to be an outlier contour trace, removing the outlier contour trace from the audio sample.

Abstract: An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal.

Abstract: Spectral encoding methods are more robust when used with improved weak signal detection and synchronizations methods. Further robustness gains are achieved by using informed embedding, error correction and embedding protocols that enable signal to noise enhancements by folding and pre-filtering the received signal.

Abstract: An audio decoder for providing at least four bandwidth-extended channel signals on the basis of an encoded representation provides first and second downmix signals on the basis of a jointly encoded representation of the first and second downmix signals using a multi-channel decoding and provides at least first and second audio channel signals on the basis of the first downmix signal using a multi-channel decoding, and provides at least third and fourth audio channel signals on the basis of the second downmix signal using a multi-channel decoding. It performs a multi-channel bandwidth extension on the basis of the first and third audio channel signals, to obtain first and third bandwidth-extended channel signals, and performs a multi-channel bandwidth extension on the basis of the second and fourth audio channel signals, to obtain second and fourth bandwidth extended channel signals. An audio encoder uses a related concept.

Abstract: The present application discloses a data processing method and a terminal thereof. The method includes: obtaining, in real time, target audio data from an on-line source; processing the target audio data using a first audio data processing approach and playing the processed target audio data at the terminal; while playing the processed target audio data: obtaining an audio data processing approach transition instruction, the audio data processing approach transition instruction including a second audio data processing approach and a real-time window of switching from the first audio data processing approach to the second audio data processing approach; in response to the audio data processing approach transition instruction, processing the target audio data received in the real-time window using the first audio data processing approach and the second audio data processing approach separately; and determining output audio data to be played at the terminal during the real-time window.

Abstract: Methods and systems are provided for optimizing call quality and improving network efficiency by reducing bandwidth requirements at the individual-voice-call level. Embodiments provided herein build vocal fingerprints that correspond to the frequency range of the human voice, as well as the frequency range of the voice of individual users. The vocal fingerprints are used minimize and reduce the transmission of background noise and ambient sound captured using HD voice while retaining the frequency range of a user's voice in HD voice. This filtered HD voice frequency range is then transmitted to recipients over the network. The reduced frequency range lowers bandwidth usage and conserves network resources, all while optimizing the call quality for individual users.

Abstract: A noise detection method and apparatus are disclosed. The noise detection method includes: obtaining a frequency-domain energy distribution parameter of a current frame of an audio signal, and obtaining a frequency-domain energy distribution parameter; obtaining a tone parameter of the current frame, and obtaining a tone parameter; determining, according to the tone parameter of the current frame and the tone parameter of each of the frames in the preset neighboring domain range of the current frame, whether the current frame is in a speech section or a non-speech section; and determining that the current frame is speech-grade noise if the current frame is in a speech section and a quantity of frequency-domain energy distribution parameters falling within a preset speech-grade noise frequency-domain energy distribution parameter interval in all the frequency-domain energy distribution parameters is greater than or equal to a first threshold.

Abstract: An audio decoder is proposed for decoding a multi-object audio signal including a downmix signal X and side information PSI. The side information includes object-specific side information PSIi for an audio object si in a time/frequency region R(tR,fR), and object-specific time/frequency resolution information TFRIi indicative of an object-specific time/frequency resolution TFRh of the object-specific side information for the audio object si in the time/frequency region R(tR,fR). The audio decoder includes an object-specific time/frequency resolution determiner 110 configured to determine the object-specific time/frequency resolution information TFRIi from the side information PSI for the audio object si. The audio decoder further includes an object separator 120 configured to separate the audio object si from the downmix signal X using the object-specific side information in accordance with the object-specific time/frequency resolution TFRIi.

Abstract: A method for discriminating and attenuating pre-echo in a digital audio signal and generated from transform coding. The method includes the following acts in which, for a current frame broken down into sub-blocks, the low-energy sub-blocks precede a sub-block in which a transition or attack is detected, and determine a pre-echo area in which a pre-echo attenuation process is carried out. In the event that an attack is detected from the sub-block of the current frame, the method includes: calculating an energy leading coefficient for at least two sub-blocks of the current frame preceding the sub-block in which an attack is detected; comparing the leading coefficient to a predefined threshold; and inhibiting the pre-echo attenuation process in the pre-echo area in the event that the calculated leading coefficient is lower than the predefined threshold. Also provided are a discrimination and attenuation device implementing the acts of the method described and a decoder including such a device.

Abstract: A method for speech enhancement in speech communication devices and more specifically in hearing aids for suppressing stationary and non-stationary background noise in the input speech signal signals is disclosed. The method uses spectral subtraction wherein the noise spectrum is updated using quantile-based estimation without voice activity detection and the quantile values are approximated by dynamic quantile tracking without involving large storage and sorting of past spectral samples. The technique permits use of a different quantile at each frequency bin for noise estimation without introducing processing overheads. The preferred embodiment uses analysis-modification-synthesis based on Fast Fourier transform (FFT) and it can be integrated with other FFT-based signal processing techniques used in the hearing aids and speech communication devices.

Abstract: A system to facilitate measuring consumer use by using codes embedded in any raw audio or video content. The original code is then layered with additional codes either by modifying the original code or by adding additional codes in between the original code. The system recognizes a code embedded into an audio and/or video signal. This can help track consumer use and viewing.

Abstract: Disclosed is a unified speech and audio coding (USAC) audio signal encoding/decoding apparatus and method for digital radio services. An audio signal encoding method may include receiving an audio signal, determining a coding method for the received audio signal, encoding the audio signal based on the determined coding method, and configuring, as an audio superframe of a fixed size, an audio stream generated as a result of encoding the audio signal, wherein the coding method may include a first coding method associated with extended high-efficiency advanced audio coding (xHE-AAC) and a second coding method associated with existing advanced audio coding (AAC).

Abstract: In a transmit method, a set of data eigenvectors that are based on a Prometheus Orthonormal Set (PONS) code construction and orthogonal to each other are stored, wherein each of the data eigenvectors is mapped to a unique multi-bit word. A pilot sequence representing a pilot eigenvector that is based on the PONS code construction and orthogonal to each of the data eigenvectors is generated. Input data is grouped into multi-bit words and ones of the data eigenvectors mapped to the multi-bit words are selected. A spread data sequence including the selected ones of the data eigenvectors and that is synchronized to the pilot sequence is generated. An acoustic signal including the synchronized pilot sequence and the spread data sequence is generated. The acoustic signal is transmitted.

Abstract: A method comprising: receiving input signals for multiple channels; and parameterizing the received input signals into parameters defining multiple different object spectra and defining a distribution of the multiple different object spectra in the multiple channels.

Abstract: An apparatus for generating a bandwidth extended audio signal from an input signal, includes a patch generator for generating one or more patch signals from the input signal, wherein the patch generator is configured for performing a time stretching of subband signals from an analysis filterbank, and wherein the patch generator further includes a phase adjuster for adjusting phases of the subband signals using a filterbank-channel dependent phase correction.

Abstract: There is provided a method of generating a signal representing the speech of a user, the method comprising obtaining a first audio signal representing the speech of the user using a sensor in contact with the user; obtaining a second audio signal using an air conduction sensor, the second audio signal representing the speech of the user and including noise from the environment around the user; detecting periods of speech in the first audio signal; applying a speech enhancement algorithm to the second audio signal to reduce the noise in the second audio signal, the speech enhancement algorithm using the detected periods of speech in the first audio signal; equalizing the first audio signal using the noise-reduced second audio signal to produce an output audio signal representing the speech of the user.

Abstract: A novel approach provides accurate estimation of the parameter a of a Fractional Fourier Transform (FrFT). A value of a may be selected for which the Wigner Distributions (WDs) of a signal-of-interest (SOI) and interference overlap as little as possible. However, instead of computing the WD for each signal, the FrFT may be computed for each WD, recognizing that the projection of the WD of a signal onto an axis ta is the energy of the FrFT along the same axis. Since the technique computes a using the SOI and a measure of the interference separately, significant improvements can be made in the estimate, especially at low signal-to-noise ratio (SNR). Once the estimate is obtained, a reduced rank filter may be applied to remove the interference, since minimum mean-square error (MMSE) approaches will again fail when using the low sample support required of non-stationary environments. The technique is not only computationally more efficient than MMSE, but far more robust as well.

Abstract: An apparatus, method and computer program for generating a wideband signal using a lowband input signal includes a processor for performing a guided bandwidth extension operation using transmitted parameters and a blind bandwidth extension operation only using derived parameters rather than transmitted parameters. To this end, the processor includes a parameter generator for generating the parameters for the blind bandwidth extension operation.

Abstract: The present invention relates to a time delay estimation device. The time delay estimation device of the present invention includes a sound signal detection unit configured to detect sound signals through a plurality of microphones, a frequency domain conversion unit configured to convert the detected sound signals into signals of a frequency domain, and a time delay estimation unit configured to estimate a time delay on the basis of a slope of a phase difference between the sound signals converted into the frequency domain.

Abstract: An embodiment of the present invention discloses a data processing method, including: twiddling input data, so as to obtain twiddled data; pre-rotating the twiddled data by using a symmetric rotate factor, where the rotate factor is a·W4L2p+1, p=0, . . . , L/2?1, and a is a constant; performing a Fast Fourier (Fast Fourier Transform, FFT) transform of L/2 point on the pre-rotated data, where L is the length of the input data; post-rotating the data that has undergone the FFT transform by using a symmetric rotate factor, where the rotate factor is b·W4L2q+1, q=0, . . . , L/2?1, and b is a constant; and obtaining output data.

Abstract: Embodiments of the present invention provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.

Abstract: Dynamic scaling of messaging content is facilitated. Messaging content can include visual content other than the actual textual content of a message body for a messaging environment. In an aspect dynamic scaling of messaging content can allow message content such as pictures, images, emoticons, movies, maps, and the like to be presented to a user in a manner that facilities efficient comprehension of the message content. The tokens can be dynamically scaled as the messaging environment changes to maintain high levels of comprehension.

Abstract: A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted.

Abstract: Disclosed are a method and apparatus for encoding and decoding a high frequency for bandwidth extension. The method includes: estimating a weight; and generating a high frequency excitation signal by applying the weight between random noise and a decoded low frequency spectrum.

Abstract: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

Abstract: The various embodiments described herein generally provide apparatus, systems and methods which facilitate the transmission of data between a client device and a remote device over a wireless telephone network. More particularly, data from a client device is modulated into an audio signal and transmitted to a wireless telephone. The wireless telephone receives the audio signal and places a phone call to a remote device over a wireless telephone network. During the phone call, the wireless telephone transmits the audio signal across a voice channel of the wireless telephone network to the remote device. The remote device receives and decodes the audio signal to extract the transmitted data.

Abstract: Techniques are described for altering the gameplay of a computer game based on imported audio content. Embodiments of the invention receive a selection of external audio content to import into a computer game. Embodiments import the selected audio content into the computer game. One or more acoustic characteristics of the imported audio content are then determined by analyzing the imported audio content. Embodiments of the invention then affect one or more graphical elements of the computer game, based on the determined one or more acoustic characteristics of the audio content.

Abstract: The present invention is provided with: a voice input section that receives a remark (a question) via a voice signal; a reply creation section that creates a voice sequence of a reply (response) to the remark; a pitch analysis section that analyzes the pitch of a first segment (e.g., word ending) of the remark; and a voice generation section (a voice synthesis section, etc.) that generates a reply, in the form of voice, represented by the voice sequence. The voice generation section controls the pitch of the entire reply in such a manner that the pitch of a second segment (e.g., word ending) of the reply assumes a predetermined pitch (e.g., five degrees down) with respect to the pitch of the first segment of the remark. Such arrangements can realize synthesis of replying voice capable of giving a natural feel to the user.

Abstract: Examples are disclosed for reproducing a voice for a speaker based on vocal tract sensing using ultra wide band (UWB) radar. These examples may include sensing a vocal tract of the speaker during non-sounded speech communication and mapping information associated with the sensed vocal tract to a voice model to generate a simulation of the vocal tract during sounded speech communication. The examples may also include reproducing a voice for the speaker based on the simulation.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: An audio correction apparatus and an audio correction method. The audio correction method includes: receiving audio data, which may be input by a user and/or an instrument uttering sounds; detecting onset information by analyzing harmonic components of the received audio data; detecting pitch information of the received audio data based on the detected onset information; comparing the audio data with reference audio data and aligning the two based on the detected onset information and the detected pitch information; and correcting the aligned audio data to match the reference audio data.

Abstract: An apparatus and method are disclosed for filtering and performing high frequency reconstruction of an audio signal. The apparatus includes an analysis filter bank, a high frequency reconstructor, and a synthesis filter bank. The analysis filterbank receives real-valued time domain input audio samples and generates complex valued subband samples. The high frequency reconstructor modifies at least some of the complex valued subband samples. The synthesis filter bank receives the modified complex valued subband samples and generates time domain output audio samples. The analysis filter bank comprises analysis filters that are complex exponential modulated versions of a prototype filter with an arbitrary phase shift.

Abstract: An audio driver with reduced rub and buzz distortion that includes a digital processing module. A digital to audio converter (DAC) operable to receive a digital audio signal supplied by the digital processing module. One or more analog driver stages operable to receive an analog audio signal supplied by the DAC. A peak amplitude compressor.

Abstract: A threshold amplitude is calculated for each subband obtained by splitting an extension band. For each subband, an amplitude of transform coefficients is compared with the threshold amplitude to extract a transform coefficient having an amplitude larger than the threshold amplitude as a representative transform coefficient. When a number of the extracted representative transform coefficients is less than a predetermined number, the threshold amplitude is updated in accordance with an amount by which the number of the representative transform coefficients is less than the predetermined number. A transform coefficient is extracted again using the updated threshold amplitude. For each of the subbands, a value of correlation is calculated between the representative transform coefficient and a normalized core encoded low-band transform coefficient. A subband having a largest value of correlation is selected when the number of the extracted representative transform coefficients reaches the predetermined number.

Abstract: Encoding and decoding methods and apparatus are described to obtain auxiliary information from an audio signal. The auxiliary information uses a plurality of frequency components residing in a plurality of code bands. The audio signal is transformed into a frequency domain representation. Characteristics of frequencies of the frequency domain representation that may contain the auxiliary information are determined. The characteristics of frequencies of the frequency domain representation in a respective one of the code bands that may contain the auxiliary information are normalized across the code band. The normalization is carried out against a characteristic of a frequency in that code band. The normalized characteristics of the frequencies representative of auxiliary information are summed to determine a sum for a frequency representative of auxiliary information. It is determined when the sum is representative of the auxiliary information.

Abstract: A parameter estimation method for inactive voice signals and a system thereof and comfort noise generation method and system are disclosed. The method includes: for an inactive voice signal frame, performing time-frequency transform on a sequence of time domain signals containing the inactive voice signal frame to obtain a frequency spectrum sequence, calculating frequency spectrum coefficients according to the frequency spectrum sequence, performing smooth processing on the frequency spectrum coefficients, obtaining a smoothly processed frequency spectrum sequence according to the smoothly processed frequency spectrum coefficients, performing inverse time-frequency transform on the smoothly processed frequency spectrum sequence to obtain a reconstructed time domain signal, and estimating an inactive voice signal parameter according to the reconstructed time domain signal to obtain a frequency spectrum parameter and an energy parameter.

Abstract: Disclosed is a communication technology in which the effect of packet loss can be easily reduced to the extent that it can be ignored, even on networks where packet loss can easily occur. Specifically, a communication device is disclosed that includes a packet loss determination unit that determines whether a packet that transmits image information has been lost, and an interpolated packet transmission unit that transmits an interpolated packet when a packet that transmits image information has been lost.

Abstract: A method and an apparatus for detecting noise of audio signals are provided. The method includes steps of converting an audio signal into a plurality of audio frames, where the audio frames are arranged in chronological order while taking a target frame as a center, calculating a plurality of magnitudes respectively corresponding to a plurality of spectral components of each of the audio frames, calculating differences between the adjacent magnitudes in a time-frequency domain to obtain a plurality of difference values in at least two directions orthogonal to each other in the time-frequency domain, where the time-frequency domain is defined by the audio frames, determining a maximum degree of difference of the magnitudes in the time-frequency domain according to the difference values, and determining whether a part of the audio signal corresponding to the target frame is a noise according to the maximum degree of difference.

Abstract: For high-accuracy analysis and high-quality synthesis of voice sound (singing and speech), provided herein are a system and a method for estimating from an audio signal spectral envelopes and group delays for sound analysis and synthesis with high accuracy and high temporal resolution. An estimation system of spectral envelopes and group delays includes a fundamental frequency estimation section, an amplitude spectrum acquisition section, a group delay extraction section, a spectral envelope integration section, and a group delay integration section. The spectral envelope integration section sequentially obtains a spectral envelope for sound synthesis by averaging overlapped spectra. The group delay integration section selects from a plurality of group delays a group delay corresponding to the maximum envelope of each frequency component of the spectral envelope and integrates groups delays thus selected to sequentially obtain a group delay for sound synthesis.

Abstract: A method for predicting a bandwidth extension frequency band signal includes demultiplexing a received bitstream to obtain a frequency domain signal; determining whether a highest frequency bin, to which a bit is allocated, of the frequency domain signal is less than a preset start frequency bin of a bandwidth extension frequency band; predicting an excitation signal of the bandwidth extension frequency band according to the determination; and predicting the bandwidth extension frequency band signal according to the predicted excitation signal of the bandwidth extension frequency band and a frequency envelope of the bandwidth extension frequency band.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: A voice processing apparatus includes: a dividing unit which divides a voice signal into frames in such a manner that any two successive frames overlap each other by a predetermined amount; a first windowing unit which multiplies each frame by a first windowing function that attenuates a signal at both ends of the frame; an orthogonal transform unit which computes a frequency spectrum for each frame multiplied by the first windowing function; a frequency signal processing unit which computes a corrected frequency spectrum; an inverse orthogonal transform unit which computes a corrected frame by applying an inverse orthogonal transform to the corrected frequency spectrum; a second windowing unit which multiplies each corrected frame by a second windowing function that attenuates a signal at both ends of the corrected frame; and an addition unit which adds up the each corrected frame multiplied by the second windowing function, sequentially in time order.

Abstract: A method for determining mantissa bit allocation of audio data values of frequency domain audio data to be encoded. The allocation method includes a step of determining masking values for the audio data values, including by performing adaptive low frequency compensation on the audio data of each frequency band of a set of low frequency bands of the audio data. The adaptive low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set of low frequency bands has prominent tonal content; and performing low frequency compensation on the audio data in each frequency band in the set of low frequency bands having prominent tonal content as indicated by the compensation control data, but not performing low frequency compensation on the audio data in any other frequency band in the set of low frequency bands.

Abstract: Examples disclosed herein collect research data relating to media received in a computer processing device. The research data includes a cookie associated with accessing the media. Examples disclosed herein present a message requesting communicative coupling of a portable computing device to the computer processing device when the portable computing device is not inductively coupled to the computer processing device. Examples disclosed herein receive information from the portable computing device via an inductive connection when it is detected that the portable computing device is coupled to the computer processing device. The information is indicative of a user of the portable computing device. The information is associated to the research data.

Abstract: A method, system, and device provide power-efficient communications within the context of available power. Transmission and receipt data rates are scalable in accordance with output power available from a power source. Data is transmitted at a data rate determined, at least in part, by the available output power.

Abstract: Embodiments of the present invention provide an audio signal coding and decoding method and device. The coding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantized sub-band normalization factor, or according to the quantized sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantized sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

Abstract: An “Overcomplete Audio Coder” provides various techniques for overcomplete encoding audio signals using an MCLT-based predictive coder. Specifically, the Overcomplete Audio Coder uses unrestricted polar quantization of MCLT magnitude and phase coefficients. Further, quantized magnitude and phase coefficients are predicted based on properties of the audio signal and corresponding MCLT coefficients to reduce the bit rate overhead in encoding the audio signal. This prediction allows the Overcomplete Audio Coder to provide improved continuity of the magnitude of spectral components across encoded signal blocks, thereby reducing warbling artifacts. Coding rates achieved using these prediction techniques are comparable to that of encoding an orthogonal representation of an audio signal, such as with modulated lapped transform (MLT)-based coders.

Abstract: In a method of improving perceived loudness and sharpness of a reconstructed speech signal delimited by a predetermined bandwidth, performing the steps of providing (S10) the speech signal, and separating (S20) the provided signal into at least a first and a second signal portion. Subsequently, adapting (S30) the first signal portion to emphasize at least a predetermined frequency or frequency interval within the first bandwidth portion. Finally, reconstructing (S40) the second signal portion based on at least the first signal portion, and combining (S50) the adapted first signal portion and the reconstructed second signal portion to provide a reconstructed speech signal with an overall improved perceived loudness and sharpness.