Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing entertainment audio, such as television audio, to improve the clarity and intelligibility of speech, such as dialog and narrative audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: The invention presents a method to improve the recovering from packet loss, frame erasure or jitter concealment during signal communication, especially for VoIP (Voice Over Internet Protocol) applications. A variable delay concept (instead of constant delay) is introduced to guarantee the continuity and periodicity of signal after recovering lost frames, adding frames or removing frames. During the recovering of lost frames or the adding of extra frames, the copy of previous signal from history buffer into missing frame(s) is based on the frame length, onset, and offset information.

Abstract: Provided are methods and systems for improving quality of speech communications. The method may be for improving quality of speech communications in a system having a speech encoder configured to encode a first audio signal using a first set of encoding parameters associated with a first noise suppressor. A method may involve receiving a second audio signal at a second noise suppressor which provides much higher quality noise suppression than the first noise suppressor. The second audio signal may be generated by a single microphone or a combination of multiple microphones. The second noise suppressor may suppress the noise in the second audio signal to generate a processed signal which may be sent to a speech encoder. A second set of encoding parameters may be provided by the second noise suppressor for use by the speech encoder when encoding the processed signal into corresponding data.

Abstract: An apparatus, electronic apparatus and method for adjusting jitter buffer is provided. A previous jitter buffer size based on a jitter buffer size determined according to an adaptive jitter buffer size calculation algorithm is applied in predicting a jitter buffer size of future time such that the predicted jitter buffer size is applied to obtain a jitter buffer size of a valid time. The audio quality of the speech transmitted over a packet switched network is enhanced.

Abstract: A system improves speech detection or processing by identifying registration signals. The system encodes a limited frequency band by varying the amplitude of a pulse width modulated signal between predefined values. The signal is separated into frequency bins that identify amplitude and phase. The registration signal is measured by comparing a difference in average acoustic power in a plurality of adjacent bins over time.

Abstract: Decoder for an audio signal coded by a coder including a long-term prediction filter wherein the decoder comprises: a block (211) for detecting transmission frame losses; a module (222) for calculating values of an error indication function representative of the cumulative adaptive excitation error during decoding following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation gain for the lost frame; a module (213) for calculating an error indication parameter from said values of the error indication function; a comparator (214) for comparing said error indication parameter to at least one given threshold; and a discriminator (215) adapted to determine as a function of the results supplied by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder.

Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.

Abstract: Various embodiments of the invention provide scalable and distributed input signal coding activity detection and coding thereof (e.g. VAD/DTX) processing framework. An apparatus comprising an encoder is shown. The apparatus can be a terminal, for example a mobile phone, computer or the like. The apparatus may act as transmitter etc. The apparatus is coupled with a network element (alternatively referred to as an intermediate element or the like). The network element is coupled with apparatuses. The apparatuses can also be terminal devices such as mobile phone, computer or the like. The apparatuses may act as receivers etc. The apparatus comprises a detector configured to detect whether an input signal is active input signal or non-active input signal. There are various different detectors such as the VAD or SAD referred to above.

Abstract: An audio decoding device performs frame loss compensation capable of obtaining a decoded audio which is natural for ears with little noise. The audio decoding device includes a non-cyclic pulse waveform detection unit for detecting a non-cyclic pulse waveform section in a n?1-th frame, which is repeatedly used with a pitch cycle in the n-th frame upon compensation of loss of the n-th frame. The audio coding device also includes a non-cyclic pulse waveform suppression unit for suppressing a non-cyclic pulse waveform by replacing an audio source signal existing in the non-cyclic pulse waveform section in the n?1-th frame by a noise signal. The audio coding device further includes a synthesis filter for using a linear prediction coefficient decoded by an LPC decoding unit to perform synthesis by a synthesis filter by using the audio source signal of the n?1-th frame from the non-cyclic pulse waveform suppression unit as a drive audio source, thereby obtaining the decoded audio signal of the n-th frame.

Abstract: An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

Abstract: The present disclosure relates to a decoding method and apparatus. The method includes: receiving data frames from the coder; if any erroneous frame appears, calculating a pitch lag parameter of the erroneous frame; decoding the data frames according to the calculated pitch lag parameter of the erroneous frame, and obtaining decoded data. The process of determining the pitch lag parameter includes: determining the number of continuous erroneous frames and the pitch lag parameter of the previous frame; adjusting the pitch lag parameter of the previous frame according to the number of the continuous erroneous frames and a preset adjustment policy, and calculating and determining the pitch lag parameter of a current erroneous frame, wherein the preset adjustment policy is adjusting the determined pitch lag parameter of the current erroneous frame within a preset value range according to the number of the continuous erroneous frames.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

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 noise canceller is constituted by a forward linear prediction processor that performs forward linear prediction for a reception signal in a noisy section where the reception signal includes noise, and a backward linear prediction processor that performs backward linear prediction for the reception signal in the noisy section. A signal generator generates a replacement signal, to replace the reception signal in the noisy section, based on the forward linear prediction values obtained by the forward linear prediction and the backward linear prediction values obtained by the backward linear prediction. The signal generator obtains replacement values for a plurality of points within the noisy section by weighting the corresponding forward prediction value and the corresponding backward prediction value with an internal division ratio, the internal division ratio having a linear or nonlinear functional relationship to the time axis of the signal.

Abstract: A transformation-parameter calculating unit calculates a first model parameter indicating a parameter of a speaker model for causing a first likelihood for a clean feature to maximum, and calculates a transformation parameter for causing the first likelihood to maximum. The transformation parameter transforms, for each of the speakers, a distribution of the clean feature corresponding to the identification information of the speaker to a distribution represented by the speaker model of the first model parameter. A model-parameter calculating unit transforms a noisy feature corresponding to identification information for each of speakers by using the transformation parameter, and calculates a second model parameter indicating a parameter of the speaker model for causing a second likelihood for the transformed noisy feature to maximum.

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: Envelope identification section generates input envelope data (DEVin) indicative of a spectral envelope (EVin) of an input voice. Template acquisition section reads out, from a storage section, converting spectrum data (DSPt) indicative of a frequency spectrum (SPt) of a converting voice. On the basis of the input envelope data (DEVin) and the converting spectrum data (DSPt), a data generation section specifies a frequency spectrum (SPnew) corresponding in shape to the frequency spectrum (SPt) of the converting voice and having a substantially same spectral envelope as the spectral envelope (EVin) of the input voice, and the data generation section generates new spectrum data (DSPnew) indicative of the frequency spectrum (SPnew). Reverse FFT section and output processing section generates an output voice signal (Snew) on the basis of the new spectrum data (DSPnew).

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: A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

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: To increase channel capacity, mobile phone carriers have deployed speech coders, such as Advanced MultiBand Excitation coding (AMBE), in networks to reduce the bit rate of each call. One undesired consequence of employing such speech coders is that the voice quality can be much worse as compared to higher bit-rate speech coders. A method or corresponding apparatus in an example embodiment of the present invention performs voice quality enhancement transparently within a network by detecting use of a coder applying rate reduction to a speech signal and known to have an adverse effect on a coded speech signal. Upon detection of the use of such coder, the coded speech signal is corrected based on components introduced into the coded speech signal due to the rate reduction. As a result of applying the voice quality enhancement, adverse effects of speech coders can be reduced, while maintaining high quality voice signals.

Abstract: A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract: A method of setting an equalizer so as to enlarge a sound field in reproducing an audio file and a method of reproducing an audio file thereby, includes: dividing an input audio file into segments with a predetermined time length; extracting an audio feature value for each segment; determining equalizer information for reproducing each segment by the use of the extracted feature value; and determining an equalizer sequence for the audio file by the use of the determined equalizer information of each segment. Since the equalizer setting information can be automatically changed without user manipulation, the user can listen to an audio file of which the sound field is dynamically enlarged.

Abstract: An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.

Abstract: Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

Abstract: A method of reducing noise in a speech signal involves converting the speech signal to the frequency domain using a fast fourier transform (FFT), creating a subset of selected spectral subbands, determining the appropriate gain for each subband, and interpolating the gains to match the number of FFT points. The converted speech signal is then filtered using the interpolated gains as filter coefficients, and an inverse FFT performed on the processed signal to recover the time domain output signal.

Abstract: To reduce audio glitch rendering buffer of an audio application is pre-filled with natural sounding audio rather than zeros. For every frame of audio sent for rendering, the rendering buffer is also pre-filled or the signal is stretched in the buffer in anticipation of a glitch. If the glitch does not occur, then the stretched signal is overwritten and the end user does not notice it. If the glitch does occur, then the rendering buffer is already filled with a stretched version of the previous audio and may result in sound that is acceptable. After recovery from the glitch, any new data is smoothly merged into the fake audio that was generated before.

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: A noise-canceling device of a voice communication terminal that removes noise elements included in received voice signals. The device comprises: a digital filter array that exhibits filter qualities in response to a coefficient setting signal showing each supplied arrays of filter coefficients, and includes a first-stage filter that receives the received voice signals as well as multiple later-stage filters connected thereto in a straight line; a filter qualities designator that generates input designation that designates each qualities of the multiple digital filters forming the digital filters array; and a filter coefficient setter that retains multiple arrays of filter coefficients, extracts a filter coefficient array corresponding to the designation input from among the multiple filter coefficient arrays, and supplies to each multiple digital filters.

Abstract: This disclosure describes signal processing techniques that can improve the performance of blind source separation (BSS) techniques. In particular, the described techniques propose pre-processing steps that can help to de-correlate the different signals from one another prior to execution of the BSS techniques. In addition, the described techniques also propose optional post-processing steps that can further de-correlate the different signals following execution of the BSS techniques. The techniques may be particularly useful for improving BSS performance with highly correlated audio signals, e.g., from two microphones that are in close spatial proximity to one another.

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: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.

Abstract: A speech processing apparatus includes a rule storing unit that stores therein a rule that correlates one another causes of errors in speech recognition, responding methods each of which is used when an error has occurred during the speech recognition, and responding users each of whom is one of a plurality of users and serving as a target of a response; a detecting unit that detects a cause of an error that has occurred during the recognition of the speech; a method selecting unit that selects one of the responding methods that is correlated with the detected cause of the error from the rule storing unit; a user selecting unit that selects one of the responding users that is correlated with the detected cause of the error from the rule storing unit; and an executing unit that executes the response by the selected responding method to the selected responding user.

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: In a speech decoding method and apparatus, an adaptive code vector is obtained from an adaptive codebook, and a time series vector is obtained from an excitation codebook. Gains of the adaptive code vector and an excitation code vector are respectively decoded from a gain code. The gain of the adaptive code vector is classified into a first gain corresponding to a first noise level or a second gain corresponding to a second noise level. A value is determined based on the classifying results, and a mathematical operation is performed on the time series vector and the determined value. The adaptive code vector and the time series vector are weighted by the decoded gains, and an excitation signal is obtained by adding the weighted adaptive code vector and the weighted time series vector. A speech is synthesized using the excitation signal and a decoded linear prediction parameter.

Abstract: A technique for performing frame erasure concealment (FEC) in a speech decoder. One or more non-erased frames of a speech signal are decoded in a block-independent manner. When an erased frame is detected, a short-term predictive filter and a long-term predictive filter are derived based on previously-decoded portions of the speech signal. A periodic waveform component is generated using the short-term predictive filter and the long-term predictive filter. A random waveform component is generated using the short-term predictive filter. A replacement frame is generated for the erased frame. The replacement frame may be generated based on the periodic waveform component, the random waveform component, or a mixture of both.

Abstract: A method and apparatus for dynamically enabling the activation and deactivation of comfort noise over a VoIP media path or channel are disclosed. The present method detects all sound levels in the media path and only activates the comfort noise in the absence of sound and when the background noise level or the telephone line noise level is low rather than only in the absence of speech.

Abstract: The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In a preferred encoder embodiment an audio signal is encoded by deterministic encoder means to form a first encoded signal part. A spectrum of the audio signal is determined and represented by an excitation pattern, i.e. spectral values corresponding to human auditory filters, as a second encoded signal part. A masking curve is also extracted based on the excitation pattern, thus improving encoding efficiency in terms of bit rate. In a preferred decoder the first encoded signal part is decoded by deterministic decoder means. A noise generator uses the decoded first signal part together with the second signal part, i.e. the excitation pattern for the original audio signal, to generate a noise signal. The noise signal is then added to the first decoded signal part to form an output audio signal. At the decoder side the masking curve is also extracted based on the second encoded signal part, i.e.

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: To reduce noise in an input signal that may contain speech, first an estimate of the noise level in the signal is obtained. The level of the input signal is then compared with the noise level estimate signal to determine whether speech is dominant. Less aggressive noise reduction is applied to the input signal when speech is dominant than when only noise is present.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: An acoustic signal enhancement method is disclosed. The acoustic signal enhancement method comprises the steps of applying a spectral transformation on a frame derived from an input acoustic signal to generate a spectral representation of the frame, estimating an a posteriori SNR and an a priori SNR of the frame, determining an a priori SNR limit for the frame, limiting the a priori SNR with the a priori SNR limit to generate a final a priori SNR for the frame, determining a spectral gain for the frame according to the a posteriori SNR and the final a priori SNR, and applying the spectral gain on the spectral representation of the frame so as to generate an enhanced spectral representation of the frame. One of the characteristics of the acoustic signal enhancement method is that the a priori SNR limit is a function of frequency.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: A method includes receiving a sequence of frames containing audio information and determining that a frame is missing in the sequence of frames. The method also includes comparing the frame that precedes the missing frame to the received frames to identify a selected frame. The method further includes identifying a replacement frame comprising the frame that follows the selected frame. In addition, the method includes inserting the replacement frame into the sequence of frames in place of the missing frame.

Abstract: In one embodiment, distortion in a received speech signal is estimated using at least one model trained based on subjective quality assessment data. A speech quality assessment for the received speech signal is then determined based on the estimated distortion.

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: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.