Abstract: Automatically normalizing a perceived loudness for a digitally encoded audio track formed of a number of channels during playback on a multimedia asset player is described. A number of auditory selectivity frequency bands are selected and for each channel in the track, a power value for each of the number of selectivity frequency bands is computed. Each of the power values is weighted by a sensitivity weighting factor and a sum value of all the weighted power values is then calculated. For the track, a perceived acoustic power value is calculated based upon the sum value for each of the channels and a normalization gain factor based upon the perceived acoustic power is calculated and associated with the track. During playback, the normalization gain factor is applied to the track.

Abstract: A method and a device are for operating a voice-enhancement system, such as a communication and/or intercom/two-way intercom or duplex telephony device in a motor vehicle. The device includes at least one microphone and at least one loudspeaker for reproducing a signal generated by the microphone, as well as a bandpass filter configured between the microphone and the loudspeaker. The bandpass filter is adjusted as a function of a comparison between the power of the signal generated by the microphone at a test frequency, and the power of the signal generated by the microphone at an at least substantially integral multiple of the test frequency, or as a function of a comparison between the power of the signal generated by the microphone at a test frequency, and the power of the signal generated by the microphone at the test frequency at at least an earlier point in time.

Abstract: A common problem in audio processing is that a useful signal is disturbed by one or more sinusoidal noises that should be suppressed. One embodiment of the invention provides a method of canceling a sinusoidal disturbance of unknown frequency in a disturbed useful signal. The method comprises the steps of estimating parameters of the sinusoidal disturbance including amplitude, phase and frequency; generating a reference signal on the basis of the estimated parameters; and subtracting the reference signal from the disturbed useful signal. According to one embodiment of the present invention, the estimation is performed by an Extended Kalman filter.

Abstract: A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively.

Abstract: A method according to the invention separates multiple audio signals recorded as a mixed signal via a single channel. The mixed signal is A/D converted and sampled. A sliding window is applied to the samples to obtain frames. The logarithms of the power spectra of the frames are determined. From the spectra, the a posteriori probabilities of pairs of spectra are determined. The probabilities are used to obtain Fourier spectra for each individual signal in each frame. The invention provides a minimum-mean-squared error metho or a soft mask method for making this determination. The Fourier spectra are inverted to obtain corresponding signals, which are concatenated to recover the individual signals.

Abstract: Disclosed is a reproducing apparatus comprising: a reproduction section to reproduce reproduction data comprising sound data and/or image data; a selection section to calculate evaluation values between a link source set for the reproduction data and each of a plurality of link destinations corresponding to the link source by a predetermined arithmetic expression based on link information of the plurality of link destinations, and to select a link destination having a highest evaluation among the evaluation values out of the plurality of link destinations; and a reproduction control section to move a reproduction point of the reproduction data reproduced by the reproduction section to a position corresponding to the link destination by linking the link source with the link destination when the reproduction point reaches a given point with respect to a position corresponding to the link source, and to instruct the reproduction section to reproduce the reproduction data.

Abstract: In a decoding apparatus (30), power compensation spectrum generation/composition units (371 to 374) adjust power of power compensation spectrums PCSP based on quantization accuracy information, normalization coefficients, gain control information, and power adjustment information. Then, power of the spectrums SP is compensated by replacing spectrums SP being equal to or smaller than a threshold with the power-adjusted power compensation spectrums PCSP, or by adding the power-adjusted power compensation spectrums PCSP to the spectrums SP.

Abstract: A received voice processing apparatus is provided, in which the received voice processing apparatus includes: a target spectrum calculation part for calculating, for each frequency band, a target spectrum on the basis of a compression ratio for a voice spectrum; a gain calculation part for calculating a gain value for amplifying the voice spectrum to the target spectrum; a filter coefficient calculation part for calculating a filter coefficient from the gain value; and a filer part for processing a received voice signal by using the filter coefficient.

Abstract: A communication pattern inducing system focuses on the propagation of topics amongst a plurality of nodes based on the text of the node rather than hyperlinks of the node. A node could represent a weblog or any other source of information such as person, a conversation, images, etc. The system utilizes a model for information diffusion, wherein the parameters of the model capture how a new topic spreads from node to node. The system further comprises a process to learn the parameters of the model based on real data and to apply the process to real (or synthetic) node data. Consequently, the system is able to identify particular individuals that are highly effective at contributing to the spread of topics.

Abstract: Modifying an audio signal comprising a plurality of channel signals is disclosed. At least selected ones of the channel signals are transformed into a time-frequency domain. The at least selected ones of the channel signals are compared in the time-frequency domain to identify corresponding portions of the channel signals that are not correlated or are only weakly correlated across channels. The identified corresponding portions of said channel signals are modified.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: A method of sending a voice message via a mobile communication device, said method involving: receiving an utterance from a user of the mobile communication device; generating a non-text representation of the received utterance; inserting the non-text representation into a body of a text message; and sending the text message over a wireless messaging channel from the mobile communication device to a recipient's device.

Abstract: A method for processing audio signal that includes receiving a plurality of digital audio signals and calculating signal levels of them. The next step is calculating smooth attenuation signal for each one of the input audio signals. The smooth attenuation signals are calculated according to the signal levels and smoothing criteria, such that the mixing of signals that include the input audio signals multiplied by their respective smooth attenuation signals is peak limited by a given threshold level.

Abstract: Primary and alternate optimization procedures are used to improve the ITU-T G.723.1 speech coding standard (the “Standard”) by replacing the Hamming window of the Standard with an optimized window, with two windows, or with two windows and an additional performance of an autocorrelation method. When two windows replace the Hamming window, at least one of which is an optimized window, generally the first is used to determine optimized unquantized LP coefficients which are used to define an optimized perceptual weighting filter, and the second is used to determine optimized unquantized LP coefficients which are used to determine optimized synthesis coefficients. Optimized windows created using the primary and alternate optimization procedures and used in the Standard yield improvements in the objective and subjective quality of synthesized speech produced by the Standard. The improved Standard, methods, and window can all be implemented as computer readable software code.

Abstract: A fault severity check and source identification method for a frequency domain instrument accesses acquired reflection data for a transmission line under test. From the acquired data reflection surfaces are isolated as a function of distance. Each reflection surface is examined to produce a frequency response profile and a worst-case reflection response to determine fault severity. The frequency response profile may also be correlated in a pattern recognition algorithm with known reference source profiles to determine the source identification for the fault.

Abstract: A system and method are disclosed for transient detection and modification in audio signals. Digital signal processing techniques are used to detect transients and modify an audio signal to enhance or suppress such transients, as desired. A transient audio event is detected in a first portion of the audio signal. A graded response to the detected transient audio event is determined. The first portion of the audio signal is modified in accordance with the graded response. The extent of enhancement or suppression (as applicable) may be determined at least in part by a measure of the significance or magnitude of the transient.

Abstract: A method to select features for maximum entropy modeling in which the gains for all candidate features are determined during an initialization stage and gains for only top-ranked features are determined during each feature selection stage. The candidate features are ranked in an ordered list based on the determined gains, a top-ranked feature in the ordered list with a highest gain is selected, and the model is adjusted using the selected top-ranked feature.

Abstract: Methods and apparatus, in the context of speech recognition, for compensating in the cepstral domain for the effect of an interfering signal by using a reference signal.

Abstract: An audio encoder regulates quality and bitrate with a control strategy. The strategy includes several features. First, an encoder regulates quantization using quality, minimum bit count, and maximum bit count parameters. Second, an encoder regulates quantization using a noise measure that indicates reliability of a complexity measure. Third, an encoder normalizes a control parameter value according to block size for a variable-size block. Fourth, an encoder uses a bit-count control loop de-linked from a quality control loop. Fifth, an encoder addresses non-monotonicity of quality measurement as a function of quantization level when selecting a quantization level. Sixth, an encoder uses particular interpolation rules to find a quantization level in a quality or bit-count control loop. Seventh, an encoder filters a control parameter value to smooth quality. Eighth, an encoder corrects model bias by adjusting a control parameter value in view of current buffer fullness.

Abstract: A method and apparatus for noise suppression is described herein. The channel gain is controlled based on a degree of variability of the background noise. The noise variability estimate is used in conjunction with a variable attenuation concept to produce a family of gain curves that are adaptively suited for a variety of combinations of long-term peak SNR and noise variability. More specifically, a measure of the variability of the background noise is used to provide an optimized threshold that reduces the occurrence of non-stationary background noise entering into the transition region of the gain curve.

Abstract: An encoding device (100) includes (i) a first encoding unit (132) that encodes spectral data in the lower frequency band represented by a plularity of parameters, out of the spectral data obtained by transforming an audio signal inputted for a fixed time length, (ii) a second quantizing unit (133) that generates sub information representing characteristics of the spectral data in the higher frequency by fewer parameters than those for the lower frequency band, out of the spectral data obtained by the transformation, (iii) a second encoding unit (134) that encodes the generated sub information, and (iv) a stream output unit (140) that outputs the data encoded by the first encoding unit (132) and the data encoded by the second encoding unit (134).

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: A method of quantizing a model parameter includes applying the model parameter to a non-linear scaling function to produce a scaled model parameter and quantizing the scaled model parameter to form a quantized model parameter. In further embodiments, likelihoods for multiple frames of input feature vectors are determined for each retrieval of quantized model parameters from memory.

Abstract: For the coding or decoding of speech signal sampled values, the values contained in the code books/code tables for the generation of the speech signal parameters are stored in quantized form. The processing can be carried out using processors with whole-number processing, without deterioration of the speech quality.

Abstract: Methods and systems for filtering synthesized or reconstructed speech are implemented. A filter based on a set of linear predictive coding (LPC) coefficients is constructed by transforming the LPC coefficients to the pseudo-cepstrum, a domain existing between LPC domain and the line spectral frequency (LSF) domain. The resulting filter can emphasize spectral frequencies associated with various formants, or spectral peaks, of an inverse transfer function relating to the LPC coefficients, and can de-emphasize spectral frequencies associated with various spectral minima, or spectral valleys, of the inverse transfer function relating to the LPC coefficients.

Abstract: An audio limiting circuit capable of satisfying frequency dependent limits and time domain constraints, is disclosed. In one illustrative embodiment, an input node receives an unattenuated input signal and a system modeling filter predicts the amount, if any, by which the sound pressure level that would be generated by an acoustic transducer in response to the unattenuated input signal, would exceed one or more predetermined limits. In that embodiment, an energy detector separates the excess predicted sound pressure level into one or more frequency bands and calculates the average acoustic energy associated with each band. A gain logic block determines an attenuation factor based on whether one or more of the predetermined limits has been exceeded and the attenuation factor values are smoothed to minimize abrupt changes to the unattenuated input signal. A delay buffer delays the unattenuated input signal values.

Abstract: A voice control method that allows vocal characteristics of a character to diversely be set in a computer game where characters are capable of voice output is provided. The voice control method comprises, converting a voice that is externally input or provided in advance, based upon attribute information on the character; and an output step for outputting the converted voice as voice of the character. According to this method, the voice produced by a character that appears in a computer game can be set in accordance with the character's characteristics and various voices for each character set by each player can be created.

Abstract: The gain smoothing method and device modify the amplitude of an innovative codevector in relation to background noise present in a previously sampled wideband signal. The gain smoothing device comprises a gain smoothing calculator for calculating a smoothing gain in response to a factor representative of voicing in the sampled wideband signal, a factor representative of the stability of a set of linear prediction filter coefficients, and an innovative codebook gain. The gain smoothing device also comprises an amplifier for amplifying the innovative codevector with the smoothing gain to thereby produce a gain-smoothed innovative codevector. The function of the gain-smoothing device improves the perceived synthesized signal when background noise is present in the sampled wideband signal.

Abstract: A method and apparatus are provided for reducing noise in a signal. Under one aspect of the invention, a correction vector is selected based on a noisy feature vector that represents a noisy signal. The selected correction vector incorporates dynamic aspects of pattern signals. The selected correction vector is then added to the noisy feature vector to produce a cleaned feature vector. In other aspects of the invention, a noise value is produced from an estimate of the noise in a noisy signal. The noise value is subtracted from a value representing a portion of the noisy signal to produce a noise-normalized value. The noise-normalized value is used to select a correction value that is added to the noise-normalized value to produce a cleaned noise-normalized value. The noise value is then added to the cleaned noise-normalized value to produce a cleaned value representing a portion of a cleaned signal.

Abstract: An adaptive noise suppression system includes an input A/D converter, an analyzer, a filter, and a output D/A converter. The analyzer includes both feed-forward and feedback signal paths that allow it to compute a filtering coefficient, which is input to the filter. In these paths, feed-forward signal are processed by a signal to noise ratio estimator, a normalized coherence estimator, and a coherence mask. Also, feedback signals are processed by a auditory mask estimator. These two signal paths are coupled together via a noise suppression filter estimator. A method according to the present invention includes active signal processing to preserve speech-like signals and suppress incoherent noise signals. After a signal is processed in the feed-forward and feedback paths, the noise suppression filter estimator then outputs a filtering coefficient signal to the filter for filtering the noise out of the speech and noise digital signal.

Abstract: An improved method and system for performing speech reception threshold testing includes calibrating one or more recorded spoken words to have substantially the same sound energy and presenting the one or more calibrated recorded spoken words to a test subject. A speech reception threshold of the test subject is measured by utilizing the one or more calibrated recorded spoken words wherein the speech reception threshold measured is indicative of a sound level at which the test subject can recognize the presented recorded spoken word or words.

Abstract: An improved method and system for performing speech intelligibility testing includes calibrating one or more recorded spoken words to have substantially the same sound energy and presenting the one or more calibrated recorded spoken words to a test subject. Speech intelligibility of the test subject is measured by utilizing the one or more calibrated recorded spoken words wherein the speech intelligibility measured is indicative of a percentage of the calibrated word or words that the test subject successfully identified.

Abstract: There are provided short term enhancement methods and systems to improve perceptual quality in reproduced speech. According to one aspect, a method of enhancing a speech signal includes processing said speech signal to generate a plurality of frames, wherein each of said plurality frames includes a plurality of subframes, coding a previous subframe of said plurality of subframes using Code-Excited Linear Prediction to generate a previous excitation signal, and applying short term enhancement on said previous excitation signal to enhance a current excitation signal for a current subframe.

Abstract: With respect to each of codes corresponding to code vectors in a code book stored in a code book storage section, an expectation degree storage section stores an expectation degree at which observation is expected when an integrated parameter with respect to a word as a recognition target is inputted. A vector quantization section vector-quantizes the integrated parameter and outputs a series of codes of a code vector which has a shortest distance to the integrated parameter.

Abstract: A method and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. Thus, the disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. Generally, the distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.

Abstract: Provided in accordance with the invention are a sound source identifying apparatus and method whereby objects as sound sources can be determined as to their locations with higher accuracy by using sound information and image information thereof and are separated from mixed sounds with certainty by using position information thereof. The sound source identifying apparatus (10) is constructed to include a sound collecting part; an imaging part; a sensing part; an image processing part; a sound processing part; and a control part.

Abstract: A method of decoding a speech signal based on a CELP (Code Excited Linear Prediction) with improvement in degradation of decoded sound quality in a noise period. The method includes the steps of: calculating a norm of an excitation vector for each fixed period in a noise period; smoothing the calculated norm using a norm obtained in a previous period; changing the amplitude of the excitation vector in the period using the calculated norm and the smoothed norm; and driving a synthesizing filter by the excitation vector with the changed amplitude.

Abstract: An audio limiting circuit capable of satisfying frequency dependent limits and time domain constraints, is disclosed. In one illustrative embodiment, an input node receives an unattenuated input signal and a system modeling filter predicts the amount, if any, by which the sound pressure level that would be generated by an acoustic transducer in response to the unattenuated input signal, would exceed one or more predetermined limits. In that embodiment, an energy detector separates the excess predicted sound pressure level into one or more frequency bands and calculates the average acoustic energy associated with each band. A gain logic block determines an attenuation factor based on whether one or more of the predetermined limits has been exceeded and the attenuation factor values are smoothed to minimize abrupt changes to the unattenuated input signal. A delay buffer delays the unattenuated input signal values.

Abstract: An encoder (60) and system (1) for processing a sequence of input vectors (y0 to yT) obtained from a speech signal. A filter (2) has both a current slowly evolving filter estimate output (6) and a previous slowly evolving filter estimate output (20). The current slowly evolving filter estimate output (6) provides vectors of current filtered estimate element values of a slowly evolving component of the sequence of input vectors (y0 to yT) and the previous slowly evolving filter estimate output (20) provides vectors of previous filtered estimate element values of the slowly evolving component of said sequence of input vectors (y0 to yT). There is also a parameter estimator (10), smoother module (17) and slowly evolving component encoder (65) that provides a digitized encoded slowly evolving component of the speech signal.

Abstract: An apparatus is provided for processing a wide dynamic range analog signal which comprises multiple components such as a signal with an in-phase component and a quadrature-phase component in for example separate I and Q data channels, wherein each channel has a dynamic range compressor stage and an operator stage which processes the compressed signals. Optionally the apparatus has a dynamic range expander stage following the operator stage. A method according to the invention involves processing I and Q information after first independently compressing the dynamic range of the signal according to a logarithmic transfer characteristic over a frequency range of interest. A mathematical operation through a F(i,q) function (corresponding to the operator stage) is performed on the compressed components, thereby producing normalized components. The operating transfer function F(i,q) cross links the data channels to effect normalization based on amplitude of information in each of the channels.

Abstract: The present invention provides an echo canceller having a holding portion which holds at least a communication frame length's worth of reception signals; a power calculation portion which calculates power for each communication frame based on the reception signals held in this holding portion; a divider which divides, by the power calculated for each communication frame, the echo elimination residue signal for each sample of the communication frame output from a subtractor; and an update portion which updates the tap coefficients of an adaptive filter according to the output signals from the divider.

Abstract: An application server generates and maintains a server-side data record, also referred to as a “brownie”, that includes application state information and user attribute information for multiple users within a single session controlled by a web-based browser. The brownie includes a session identifier that uniquely identifies the session, and a subsession identifier that uniquely identifies each corresponding user of the application session. As each new user is added to the session, for example by initiating a call to the new user, the application server stores the subsession identifier and corresponding application state information for the new user in the same brownie. In response to receiving a second web page request from the browser that includes the session identifier, the application server initiates a new web application instance, and recovers the brownie from the memory based on the session identifier included in the second page request.

Abstract: In a coding procedure, coding parameters are selected for coding the speech signal to achieve enhanced perceptual quality of reproduced speech. At least one coding parameter value or preferential coding parameter value is selected to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

Abstract: A decoding processing portion 11 of a speech decoder 10 is provided with an emphasis processing portion 15 for performing an emphasis process on signals to be processed (excited signals) SPC generated from coded speech signals BS. A counter portion 17 counts the number of times code errors occurred in successive frames of the coded speech signal BS, and outputs the successive frame error number. When the successive frame error number outputted form the counter portion 17 is less than or equal to a preset reference successive frame error number, a first switch SW1 and second switch SW2 are set to an emphasis processing portion 15 side. Accordingly, the signals to be processed SPC generated from various parameters included in the coded speech signals are supplied through the switch SW1 to the emphasis processing portion 15 of the decoding processing portion 11 to perform an emphasis process.

Abstract: A signal processing system is well suited for conditioning a speech signal prior to coding the speech signal to achieve enhanced perceptual quality of reproduced speech. The signal processing system may be incorporated into mobile or portable wireless communications devices, wireless infrastructure equipment, or both. The signal processing system includes a filtering arrangement for filtering an input speech signal to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

Abstract: Speaker recognition (identification and/or verification) methods and systems, in which speech models for enrolled speakers consist of sets of feature vectors representing the smoothed frequency spectrum of each of a plurality of frames and a clustering algorithm is applied to the feature vectors of the frames to obtain a reduced data set representing the original speech sample, and wherein the adjacent frames are overlapped by at least 80%. Speech models of this type model the static components of the speech sample and exhibit temporal independence. An identifier strategy is employed in which modelling and classification processes are selected to give a false rejection rate substantially equal to zero. Each enrolled speaker is associated with a cohort of a predetermined number of other enrolled speakers and a test sample is always matched with either the claimed identity or one of its associated cohort.

Abstract: A signal processing apparatus which is capable of reducing the occurrence of noise at editing points without the need to decode an encoded audio signal when editing the encoded audio signal. A recording/reproduction section reproduces an audio signal sequence including an audio signal encoded according to normalization information for controlling amplitude of the audio signal, and the normalization information. A normalization information converter changes the normalization information of the reproduced audio signal sequence.

Abstract: A perceptual weighting device for producing a perceptually weighted signal in response to a wideband signal comprises a signal pre-emphasis filter, a synthesis filter calculator, and a perceptual weighting filter. The signal pre-emphasis filter enhances the high frequency content of the wideband signal to thereby produce a pre-emphasized signal. The signal pre-emphasis filter has a transfer function of the form: P(z)=1−&mgr;z−1, wherein &mgr; is a pre-emphasis factor having a value located between 0 and 1. The synthesis filter calculator is responsive to the pre-emphasized signal for producing synthesis filter coefficients. Finally, the perceptual weighting filter processes the pre-emphasized signal in relation to the synthesis filter coefficients to produce the perceptually weighted signal. The perceptual weighting filter has a transfer function, with fixed denominator, of the form: W (z)=A (z/&ggr;1)/(1−&ggr;2z−1) where 0<&ggr;2<&ggr;1≦1.

Abstract: Different dynamic range control values are applied to the 2-channel and m-channel outputs without repeating the inverse transform or the windowing of the audio samples. First, m-channel dynamic range control values are applied to audio samples in the frequency domain (“frequency samples” or “frequency coefficients”). The frequency samples are then inverse transformed to generate audio samples in the time domain (“time samples”) and windowed to generate windowed time samples. The windowed time samples are saved and the 2-channel dynamic range control values are applied to the windowed time samples. 2-channel dynamic range control values include 2-channel scale factors that, when multiplied with groups of the windowed time samples, at least partially remove the effects of windowing and the m-ch dynamic range control values applied in the frequency domain and readjust the dynamic range for 2-channel output.

Abstract: Different dynamic range control values are applied to the 2-channel and m-channel outputs without repeating the inverse transform of the audio samples. First, m-channel dynamic range control values are applied to audio samples in the frequency domain (“frequency samples” or “frequency coefficients”). The frequency samples are then inverse transformed to generate audio samples in the time domain (“time samples”). The time samples are duplicated to two sets where the 2-channel dynamic range control values are applied to one set of time samples. 2-channel dynamic range control values include 2-channel final scales that, when multiplied with the first set of time samples, at least partially remove the effects of the m-channel dynamic range control and readjust the dynamic range for 2-channel output. The first set and the second set are then windowed.