Abstract: A method and an apparatus for transmitting a speech signal are provided. A speech signal transmitter includes a quadrature mirror filter, a base sub-band encoder, an enhancement sub-band encoder, and a network connector. The quadrature mirror filter receives a speech signal, divides the speech signal into an enhancement band speech signal and a base band speech signal, and outputs the enhancement band speech signal and the base band speech signal. The base sub-band encoder receives and encodes the base band speech signal. The enhancement sub-band encoder receives and encodes the enhancement band speech signal. The network connector multiplexes the encoded enhancement band speech signal and the encoded base band speech signal based on the kinds of networks over which speech signals are transmitted, and transmits the multiplexed signals to the networks. A speech signal is multiplexed and transmitted by various methods based on the kinds of networks. Thus, the speech signal can be efficiently transmitted.

Abstract: An audio signal generated by a device based on audio input from a user may be received. The audio signal may include at least a user audio portion that corresponds to one or more user utterances recorded by the device. A user speech model associated with the user may be accessed and a determination may be made background audio in the audio signal is below a defined threshold. In response to determining that the background audio in the audio signal is below the defined threshold, the accessed user speech model may be adapted based on the audio signal to generate an adapted user speech model that models speech characteristics of the user. Noise compensation may be performed on the received audio signal using the adapted user speech model to generate a filtered audio signal with reduced background audio compared to the received audio signal.

Abstract: A method is provided for multi-pass echo residue detection. The method includes detecting audio data, and determining whether the audio data is recognized as speech. Additionally, the method categorizes the audio data recognized as speech as including an acceptable level of residual echo, and categorizes categorizing unrecognizable audio data as including an unacceptable level of residual echo. Furthermore, the method determines whether the unrecognizable audio data contains a user input, and also determines whether a duration of the user input is at least a predetermined duration, and when the user input is at least the predetermined duration, the method extracts the predetermined duration of the user input from a total duration of the user input.

Abstract: Embodiments of the invention provide a communication device and methods for generating enhanced audio signals. An audio signal comprising a speech signal and a noise signals is acquired at the communication device. A noise processor of the communication device detects a pitch estimation of the audio signal. Thereafter, the audio signal is processed based on the pitch estimation and processing parameters of the audio signals to remove noise signals and generate an enhanced audio signal.

Abstract: A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

Abstract: An audio signal generated by a device based on audio input from a user may be received. The audio signal may include at least a user audio portion that corresponds to one or more user utterances recorded by the device. A user speech model associated with the user may be accessed and a determination may be made background audio in the audio signal is below a defined threshold. In response to determining that the background audio in the audio signal is below the defined threshold, the accessed user speech model may be adapted based on the audio signal to generate an adapted user speech model that models speech characteristics of the user. Noise compensation may be performed on the received audio signal using the adapted user speech model to generate a filtered audio signal with reduced background audio compared to the received audio signal.

Abstract: A method determines a bias reduced noise and interference estimation in a binaural microphone configuration with a right and a left microphone signal at a time-frame with a target speaker active. The method includes a determination of the auto power spectral density estimate of the common noise formed of noise and interference components of the right and left microphone signals and a modification of the auto power spectral density estimate of the common noise by using an estimate of the magnitude squared coherence of the noise and interference components contained in the right and left microphone signals determined at a time frame without a target speaker active. An acoustic signal processing system and a hearing aid implement the method for determining the bias reduced noise and interference estimation. The noise reduction performance of speech enhancement algorithms is improved by the invention. Further, distortions of the target speech signal and residual noise and interference components are reduced.

Abstract: An signal processing apparatus, system and software product for audio modification/substitution of a background noise generated during an event including, but not be limited to, substituting or partially substituting a noise signal from one or more microphones by a pre-recorded noise, and/or selecting one or more noise signals from a plurality of microphones for further processing in real-time or near real-time broadcasting.

Abstract: The invention proposes extracting one or more speech signals (151-154) as well as one or more ambient signals (131) from sound signals captured by microphones, wherein each of the speech signals corresponds to a different speaker. The invention proposes to transmit both the one or more speech signals (151-154) and the one or more ambient signals (131) to a rendering side, as opposed to sending only speech signals. This enables to reproduce the speech and ambient signals in a spatially different way at the rendering side. By reproducing the ambient signals a feeling of “being together” is created. In an embodiment, the invention enables reproducing two or more speech signals spatially from each other and from the ambient signals so that speech intelligibility is increased despite the presence of the ambient signals.

Abstract: The embodiments of the present invention relates to a voice activity detector and a method thereof. The voice activity detector is configured to detect voice activity in a received input signal comprising an input section configured to receive a signal from a primary voice detector of said VAD indicative of a primary VAD decision and at least one signal from at least one external VAD indicative of a voice activity decision from the at least one external VAD, a processor configured to combine the voice activity decisions indicated in the received signals to generate a modified primary VAD decision, and an output section configured to send the modified primary VAD decision to a hangover addition unit of said VAD.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: One aspect of the invention provides a method for enhancing speech output by an electro-acoustical transducer in a noisy listening environment. In some embodiments, this method includes: filtering an input audio signal x(t) using a filter H(z) to produce a filtered audio signal x(t) formula (I), wherein x(t) formula (I)—H(z)x(t); providing to an electro-acoustical transducer a signal corresponding to the filtered audio signal x(t) formula (I) to produce a sound wave corresponding to the filtered audio signal; and prior to filtering the audio signal using the filter, configuring the filter such that, with respect to one or more frequencies, the filtered audio signal has a higher signal level than the input audio signal, and such that the overall signal level of the filtered audio signal (slƒ) is substantially related to the overall signal level of the input signal (slr) such that si/=sllx c.

Abstract: An audio quality feedback system and method is provided. The system receives audio from a client via a communication device such as a microphone, The audio quality feedback system compares the received audio to one or more parameters regarding the quality of the feedback. The parameters include, for example, clipping, periods of silence, signal to noise ratios. Based on the comparison, feedback is generated to allow adjustment of the communication device or use of the communication device to improve the quality of the audio.

Abstract: In prediction of a speech quality evaluation score such as a phone speech, even when a background noise exists, a subjective opinion score is predicted with high precision. A speech quality evaluation system that outputs a predicted value of the subjective opinion score for an evaluation speech such as a far-end speech of a phone, includes a speech distortion calculation unit conducts, after calculating frequency characteristics of the evaluation speech, a process of subtracting given frequency characteristics from frequency characteristics of the evaluation speech, and calculates the speech distortion on the basis of the frequency characteristics after the subtracting process has been conducted, and a subjective evaluation prediction unit that calculates the predicted value of the subjective opinion score on the basis of the speech distortion.

Abstract: Embodiments of the present invention provide a method, system and peer apparatus for implementing multi-channel voice mixing, which belongs to a network communication field. The method includes: obtaining, by each peer, voice mixing quality parameters of super peers which are determined from peers according to information processing abilities of the peers; obtaining, by peers with voice input in the peers, priorities of the super peers according to the voice mixing quality parameters, and selecting at least one super peer for voice mixing from all the super peers according to the priorities of the super peers; mixing, by the at least one super peer for voice mixing, audio data of each peer with voice input; and publishing mixed data. The present invention selects a super peer to replace the existing server for implementing multi-channel voice mixing and publishing mixed data. Thus, server costs and bandwidth resources can be saved.

Abstract: A maximum-kurtosis, distortionless response (MKDR) technique and an extension, the maximum-kurtosis, Wiener estimate (MKWE) technique, are provided. In one form, blind estimates of the speech source's channel response are made from the microphone data and MVDR is applied. The source direction is estimated by finding weights that maximize output kurtosis, or the fourth central statistical moment, in the frequency domain. The MKWE approach approximates the Wiener filter by using MKDR-output noise power estimates to compute a Wiener post-filter. These approaches can be extended to block-adaptive versions if the speech source is not quickly moving in space.

Abstract: A speech detection method is presented, which includes the following steps. A first voice captured device samples a first signal and a second voice captured device samples a second signal. The first voice captured device is closer to a speech signal source than the second voice captured device. A first energy corresponding to the first signal within an interval is calculated, a second energy corresponding to the second signal within the interval is calculated, and a first ratio is calculated according to the first energy and the second energy. The first ratio is transformed into a second ratio. A threshold value is set. It is determined whether the speech signal source is detected by comparing the second ratio and the threshold value.

Abstract: The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.

Abstract: The present invention relates to a method as well as to a computing device (20) for editing a noise-database (13) containing noise information, said noise information being derived from noise signals within an audio stream (19). In order to enhance possibilities to create and utilize context information which emerge from tracking noise signals from an audio stream, for example a telephone call, the above method is characterized by the following steps: A) in a localizing step (14), determining geographical data of the location the noise signals origin from; B) in an analyzing step (15), analyzing the noise signals with reference to the noise content; C) in a linking step, linking the analyzed noise signals to said geographical data to create noise information; D) in a storing step, storing said noise information within said noise-database (13).

Abstract: An audio processing system includes a speech detector that receives and processes an audio input signal to determine if the input signal includes components indicative of speech, and provides a control signal indicative of whether or not the audio input signal includes speech. A speech processing device receives the audio input signal and processes the audio input signal to improve its quality if the control signal indicates that the audio input signal includes speech.

Abstract: A novel beamforming post-processor technique with enhanced noise suppression capability. The present beam forming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction resulting in minimal artifacts and musical noise.

Abstract: A system is configured to facilitate bidirectional voice communication between a number of data and/or telephony devices.

Abstract: A system and method for using bi-directional conversation data to improve signal presence detection are disclosed. The detector module is adapted to communicate with a signal enhancement module. The detector module collects data from a transmit direction of the connection and a receive direction of a data connection. The collected data from the transmit and the receive direction is used to classify at least one of data in the transmit direction and data in the receive direction. Responsive to the classification, the signal enhancement module enhances data in one of the transmit direction and the receive direction. Hence, data classification accuracy is improved by using data from both the transmit and receive directions. In one embodiment, the detector module applies a voice activity detection module (VAD) process to detect the presence or absence of voice data in the collected data.

Abstract: Methods and systems of identifying speech sound features within a speech sound are provided. The sound features may be identified using a multi-dimensional analysis that analyzes the time, frequency, and intensity at which a feature occurs within a speech sound, and the contribution of the feature to the sound. Information about sound features may be used to enhance spoken speech sounds to improve recognizability of the speech sounds by a listener.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

Abstract: Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.

Abstract: A voice input device, a method for manufacturing the same, and an information processing system are provided. The voice input device has a function of removing a noise component and includes a first microphone 710-1 that includes a first vibrating membrane, a second microphone 710-2 that includes a second vibrating membrane, and a differential signal generation section 720 that generates a differential signal that represents a difference between a first voltage signal and a second voltage signal. The first and second vibrating membranes are disposed so that a noise intensity ratio is smaller than an input voice intensity ratio that represents the ratio to intensity of an input voice component.

Abstract: A method and apparatus for performing speech quality assessment in a speech communication system (such as, for example, a VoIP communication system) which detects and measures the presence of impulsive noise is provided. Specifically, in one illustrative embodiment, an autoregressive (AR) model of speech (and, in particular, of the excitation of the vocal tract) is advantageously employed to estimate a short-term variance of the speech excitation, and the standard deviation of the speech excitation (i.e., the square root of the variance) is then advantageously compared to a predetermined threshold to identify whether impulsive noise is present. Then, based on a statistic analysis of any such identified impulsive noise, a speech quality assessment is generated.

Abstract: The present invention provides a voice coder for voice communication that employs a multi-microphone system as part of an improved approach to enhancing signal quality and improving the signal to noise ratio for such voice communications, where there is a special relationship between the position of a first microphone and a second microphone to provide the communication device with certain advantageous physical and acoustic properties. In addition, the communication device can have certain physical characteristics, and design features. In a two microphone arrangement, the first microphone is located in a location directed toward the speech source, while the second microphone is located in a location that provides a voice signal with significantly lower signal-to-noise ratio (SNR).

Abstract: A method of automated speech recognition in a vehicle. The method includes receiving audio in the vehicle, pre-processing the received audio to generate acoustic feature vectors, decoding the generated acoustic feature vectors to produce at least one speech hypothesis, and post-processing the at least one speech hypothesis using pitch to improve speech recognition accuracy. The speech hypothesis can be accepted as recognized speech during post-processing if pitch is present in the received audio. Alternatively, a pitch count for the received audio can be determined, N-best speech hypotheses can be post-processed by comparing the pitch count to syllable counts associated with the speech hypotheses, and the speech hypothesis having a syllable count equal to the pitch count can be accepted as recognized speech.

Abstract: A data stream is filtered to produce a filtered data stream. The data stream is analyzed based on an acoustic parameter to determine whether a predetermined condition is satisfied. At least one extraneous portion of the data stream, in which the predetermined condition is satisfied, is determined. Thereafter, the at least one extraneous portion is deleted from the data stream to produce the filtered data stream.

Abstract: A method, apparatus, and computer-readable medium for editing a data stream based on a corpus are provided. The data stream includes stream words. A sequence includes a predetermined number of sequential words of the stream words. The method, apparatus, and computer-readable medium determine whether the sequence exists in the corpus at least at a predetermined minimum frequency. When the sequence exists in the corpus at least at the predetermined minimum frequency, the sequence is edited in the data stream.

Abstract: The present invention relates to a method and a filter design apparatus for designing a digital filter arrangement for noise suppression of a signal representing an acoustic recording. The method comprises determining a desired frequency response of the digital filter arrangement. The method is characterised by including a combination of a high pass filter and a noise suppression filter in the filter arrangement. The combination of the high pass filter and the noise suppression filter is selected based on the determined desired frequency response.

Abstract: A method of analyzing time coherence in the noisy signal including the steps of: a) determining a reference signal from the noisy signal by applying treatment (10, 18) to the noisy signal that is suitable for attenuating speech components more strongly than the noise component, in particular by an adaptive recursive predictive algorithm of the LMS type; b) determining (24) a probability of speech being present/absent on the basis of the respective energy levels in the spectral domain of the noisy signal and of the reference signal; and c) deriving (26) a denoised estimate of the speech signal from the noise signal as a function of the probability of the speech being present/absent as determined in this way.

Abstract: The perceived quality of a speech signal is improved by estimating the average power of first and second signal components and applying a first gain factor to the second signal components to generate adjusted second signal components. The first gain factor is selected such that on application of the first gain factor to the second signal components, the ratio of the average power of the first signal components to the average power of the adjusted second signal components would be a first predetermined value, the first predetermined value being such as to inhibit perceptual distortion of the improved speech signal.

Abstract: The perceived quality of a narrowband speech signal truncated from a wideband speech signal is improved by generating in a third frequency band third speech components matching first speech components in a first frequency band of the narrowband signal, and generating in a fourth frequency band fourth speech components matching second speech components in a second frequency band of the narrowband signal. A first gain factor is applied to the third speech components to generate adjusted third speech components, and a second gain factor is applied to the fourth speech components to generate adjusted fourth speech components, the gain factors being selected such that the ratios of the average powers of the adjusted third and fourth speech components to the average power of the first speech components are predetermined values.

Abstract: The perceived quality of a speech signal output from a user apparatus is improved by storing ambient noise profiles each indicating a model power distribution of a respective ambient noise type as a function of frequency; the ambient noise profile at the user apparatus is measured, the measured ambient noise profile is correlated with each of the stored ambient noise profiles, the stored ambient noise profile is selected with which the measured ambient noise profile is most highly correlated, and the speech signal is manipulated in dependence on which of the stored ambient noise profiles is selected, so as to form an improved speech signal.

Abstract: A processed component calculating unit 14 obtains a transformed noise suppressed spectrum 18a based on the ratio between a noise suppressed spectrum 18 and an estimated noise spectrum 17, and a phase disturbing unit 15 performs phase disturbance to obtain a processed spectrum 19 consisting of smoothed components that make deterioration components in the noise suppressed spectrum 18 subjectively imperceptible. A signal addition unit 11 adds the processed spectrum 19 to the frequency components of the noise suppressed spectrum 18 deteriorated through the noise suppression of a noise suppressing unit 3 to suppress the deterioration components.

Abstract: A method of removing noise includes detecting a frequency spectrum of a noise signal around the transmitting terminal, when an input signal which is a mixture of a voice signal and the noise signal is received, detecting a frequency spectrum of the input signal and an energy level of the voice signal, multiplying the frequency spectrum of the noise signal by a weight value that is determined based on the energy level of the voice signal to obtain a weighted noise frequency spectrum, and subtracting the weighted noise frequency spectrum from the frequency spectrum of the input signal.

Abstract: In a communications system that demultiplexes user data words into multiple sub-words for encoding and decoding within different subword-processing paths, the minimum distance between bit errors in an extrinsic codeword can be increased by having corresponding interleavers/deinterleavers in the different subword-processing paths use different interleaving/deinterleaving algorithms.

Abstract: Accepting the speech having the noise superimposed thereon and converting it into a signal on a time axis of the speech, an amplitude component of a speech for each predetermined frequency band of the converted signal on the frequency axis is calculated. Calculating a noise reduction coefficient, the noise component is reduced by multiplying the signal on the frequency axis of the original signal by the calculated noise reduction coefficient. By estimating the target value of the remaining noise for each frequency band, a signal on a frequency axis in which a signal corresponding to a frequency band of which target value estimated by the noise target value is larger than the value of the amplitude component of the signal on the frequency axis of which noise component is reduced is corrected to a signal corresponding to the target value is restored, into a signal on a time axis.

Abstract: Techniques are described herein that provide multi-channel noise suppression based on a Teager energy ratio. A Teager energy ratio is a ratio of an average Teager energy operator (TEO) energy of a first signal to an average TEO energy of a second signal. The average TEO energy of a signal is defined by the equation: E _ signal = 1 N ? ? i = 1 N ? [ x 2 ? ( n ) - x ? ( n + 1 ) ? x ? ( n - 1 ) ] . In this equation, ?signal represents the average TEO energy of the signal; N represents the number of frames in the signal; x(n) represents a magnitude of the signal with respect to an nth frame; x(n+1) represents a magnitude of the signal with respect to an (n+1)th frame; and x(n?1) represents a magnitude of the signal with respect to an (n?1)th frame.

Abstract: A method and apparatus for removing signal noise using multiple bands are provided. The noise removal apparatus may divide the entire frequency band into a plurality of sub-bands using a multiband filter that has characteristics similar to an auditory system of a human being and may effectively remove noise in each of the sub-bands according to a frequency subtraction scheme.

Abstract: A model-based distortion compensating noise reduction apparatus for speech recognition, includes: a speech absence probability calculator for calculating the probability distribution for absence and existence of a speech using the sound absence and existence information for the frames; a noise estimation updater for estimating a more accurate noise component by updating the variance of the clean speech and noise for each frame; and a speech absence probability-based noise filter for outputting a first clean speech through the speech absence probability transmitted from the speech absence probability calculator and a first noise filter. Further, the model-based distortion compensating noise reduction apparatus includes a post probability calculator for calculating post probabilities for mixtures using a GMM containing a clean speech in the first clean speech; and a final filter designer for forming a second noise filter and outputting an improved final clean speech signal using the second noise filter.

Abstract: A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.

Abstract: A voice activity detector (100) includes a frame divider (201) for dividing frames of an input signal into consecutive sub-frames, an energy level estimator (202) for estimating an energy level of the input signal in each of the consecutive sub-frames, a noise eliminator (203) for analyzing the estimated energy levels of sets of the sub-frames to detect and eliminate from enhancement noise sub-frames and to indicate remaining sub-frames as speech sub-frames, and an energy level enhancer (205) for enhancing the estimated energy level for each of the indicated speech sub-frames by an amount which relates to a detected change of the estimated energy level for a current speech sub-frame relative to that for neighboring speech sub-frames.

Abstract: An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution.

Abstract: In one embodiment the present invention includes a method of improving audibility of speech in a multi-channel audio signal. The method includes comparing a first characteristic and a second characteristic of the multi-channel audio signal to generate an attenuation factor. The first characteristic corresponds to a first channel of the multi-channel audio signal that contains speech and non-speech audio, and the second characteristic corresponds to a second channel of the multi-channel audio signal that contains predominantly non-speech audio. The method further includes adjusting the attenuation factor according to a speech likelihood value to generate an adjusted attenuation factor. The method further includes attenuating the second channel using the adjusted attenuation factor.

Abstract: The method comprises the following steps in the frequency domain: a) combining signals into a noisy combined signal; b) estimating a pseudo-steady noise component; c) calculating a probability of transients being present in the noisy combined signal; d) estimating a main arrival direction of transients; e) calculating a probability of speech being present on the basis of a three-dimensional spatial criterion suitable for discriminating amongst the transients between useful speech and lateral noise; and f) selectively reducing noise by applying a variable gain specific to each frequency band and to each time frame.

Abstract: A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.