Abstract: The present invention discloses a method and a device for pulse encoding, and a method and a device for pulse decoding. The method for pulse encoding includes: calculating an index value of an input pulse; selecting an adjustment threshold value according to the number of pulses, and comparing the index value of the pulse with the adjustment threshold value; if the index value is smaller than the adjustment threshold value, adopting the first number of encoding bits to encode the index value, if the index value is not smaller than the adjustment threshold value, adopting the second number of encoding bits to encode the index value plus an offset value, where the first number is smaller than the second number, the first number and the second number are both positive integers, and the offset value is greater than or equal to the adjustment threshold value.

Abstract: A system enhances the quality of a digital speech signal that may include noise. The system identifies vocal expressions that correspond to the digital speech signal. A signal-to-noise ratio of the digital speech signal is measured before a portion of the digital speech signal is synthesized. The selected portion of the digital speech signal may have a signal-to-noise ratio below a predetermined level and the synthesis of the digital speech signal may be based on speaker identification.

Abstract: One provides (101) a digital audio signal having a corresponding signal bandwidth, and then provides (102) an energy value that corresponds to at least an estimate of out-of-signal bandwidth energy as corresponds to that digital audio signal. One then uses (103) the energy value to simultaneously determine both a spectral envelope shape and a corresponding suitable energy for the spectral envelope shape for out-of-signal bandwidth content as corresponds to the digital audio signal. By one approach, if desired, one then combines (104) (on, for example, a frame by frame basis) the digital audio signal with the out-of-signal bandwidth content to provide a bandwidth extended version of the digital audio signal to be audibly rendered to thereby improve corresponding audio quality of the digital audio signal as so rendered.

Abstract: An audio signal processing system including a time-frequency conversion unit which converts an audio signal in time domain into frequency domain in frame units so as to calculate a frequency spectrum of the audio signal, a spectral change calculation unit which calculates an amount of change between a frequency spectrum of a first frame and a frequency spectrum of a second frame before the first frame based on the frequency spectrum of the first frame and the frequency spectrum of the second frame, and a judgment unit which judges the type of the noise which is included in the audio signal of the first frame in accordance with the amount of spectral change.

Abstract: The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.

Abstract: Methods, systems, and computer readable storage medium related to operating an intelligent digital assistant are disclosed. A user request is received, the user request including at least a speech input received from a user. The user request including the speech input is processed to obtain a representation of user intent for identifying items of a selection domain based on at least one selection criterion. A prompt is provided to the user, the prompt presenting two or more properties relevant to items of the selection domain and requesting the user to specify relative importance between the two or more properties. A listing of search results is provided to the user, where the listing of search results has been obtained based on the at least one selection criterion and the relative importance provided by the user.

Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

Abstract: An information processing apparatus includes an acquisition unit configured to acquire a first sound recorded from a first recording apparatus and a second sound recorded from a second recording apparatus that is different from the first recording apparatus, a determination unit configured to determine a frequency band representing a voice by analyzing a frequency of the first sound, and a change unit configured to, from among frequency components representing the second sound, change a frequency component in the frequency band.

Abstract: An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder. A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.

Abstract: Background noise is modeled from an input signal comprising a desired signal and a plurality of undesired signals. At least one of the signals that comprise the input is processed to generate a signal-to-noise ratio. An articulation index is generated for the at least one of the signals that is processed. A spectrum of a speech segment is generated to improve intelligibility and quality of the speech segment based on the articulation index. A shaping logic may adjust the spectrum of the speech segment based on a comparison of the articulation index to a plurality of predetermined thresholds. Modeling of the background noise comprises modeling a tilt of the background noise.

Abstract: Certain embodiments of the present invention employ targeted speech detection as part of end-of-hold detection in an end-of-hold notification system. The targeted speech detector is configured to be particularly sensitive to specific words or phrases so as to increase the likelihood of detecting a correct end-of-hold condition while reducing the likelihood of false end-of-hold detection. Targeted speech detection may be used along with other detection mechanisms such as DTMF detection and/or background noise detection.

Abstract: A decoder arrangement comprising a receiver input for parameters of frame-based coded signals and a decoder arranged to provide frames of decoded audio signals based on the parameters. The receiver input and/or the decoder is arranged to establish a time difference between the occasion when parameters of a first frame is available at the receiver input and the occasion when a decoded audio signal of the first frame is available at an output of the decoder, which time difference corresponds to at least one frame. A postfilter is connected to the output of the decoder and to the receiver input. The postfilter is arranged to provide a filtering of the frames of decoded audio signals into an output signal in response to parameters of a respective subsequent frame.

Abstract: A noise suppressor for altering a speech signal is trained based on a speech recognition system. An objective function can be utilized to adjust parameters of the noise suppressor. The noise suppressor can be used to alter speech signals for the speech recognition system.

Abstract: Disclosed are methods and corresponding systems for audio processing of audio signals after applying a noise reduction procedure such as noise cancellation and/or noise suppression, according to various embodiments. A method may include calculating spectral envelopes for corresponding samples of an initial audio signal and the audio signal transformed by application of the noise cancellation and/or suppression procedure. Multiple spectral envelope interpolations may be calculated between these two spectral envelopes. The interpolations may be compared to predetermined reference spectral envelopes associated with predefined clean reference speech. One of the generated interpolations, which is the closest to one of the predetermined reference spectral envelopes, may be selected. The selected interpolation may be used for restoration of the transformed audio signal such that at least a part of the frequency spectrum of the transformed audio signal is modified to the levels of the selected interpolation.

Abstract: The invention relates to transform coding/decoding of a digital audio signal represented by a succession of frames, using windows of different lengths. For the coding within the meaning of the invention, it is sought to detect (51) a particular event, such as an attack, in a current frame (Ti); and, at least if said particular event is detected at the start of the current frame (53), a short window (54) is directly applied in order to code (56) the current frame (Ti) without applying a transition window. Thus, the coding has a reduced delay in relation to the prior art. In addition, an ad hoc processing is applied during decoding in order to compensate for the direct passage from a long window to a short window during coding.

Abstract: The invention relates to a method for outputting a speech signal. Speech signal frames are received and are used in a predetermined sequence in order to produce a speech signal to be output. If one speech signal frame to be received is not received, then a substitute speech signal frame is used in its place, which is produced as a function of a previously received speech signal frame. According to the invention, in the situation in which the previously received speech signal frame has a voiceless speech signal, the substitute speech signal frame is produced by means of a noise signal.

Abstract: A system improves speech intelligibility by reconstructing speech segments. The system includes a low-frequency reconstruction controller programmed to select a predetermined portion of a time domain signal. The low-frequency reconstruction controller substantially blocks signals above and below the selected predetermined portion. A harmonic generator generates low-frequency harmonics in the time domain that lie within a frequency range controlled by a background noise modeler. A gain controller adjusts the low-frequency harmonics to substantially match the signal strength to the time domain original input signal.

Abstract: Configurations disclosed herein include systems, methods and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Example embodiments may first remove any existing context from a digital audio signal to obtain a context suppressed signal. The context suppressed signal may then be encoded. An audio context may be selected from among a plurality of audio contexts, with the selected audio context inserted into a signal based on the encoded context suppressed signal.

Abstract: Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.

Abstract: A vocoder and method transcodes Mixed Excitation Linear Prediction (MELP) encoded data for use at different speech frame rates. Input data is converted into MELP parameters such as used by a first MELP vocoder. These parameters are buffered and a time interpolation is performed on the parameters with quantization to predict spaced points. An encoding function is performed on the interpolated data as a block to produce a reduction in bit-rate as used by a second MELP vocoder at a different speech frame rate than the first MELP vocoder.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: Described is a multiple phase process/system that combines spatial filtering with regularization to separate sound from different sources such as the speech of two different speakers. In a first phase, frequency domain signals corresponding to the sensed sounds are processed into separated spatially filtered signals including by inputting the signals into a plurality of beamformers (which may include nullformers) followed by nonlinear spatial filters. In a regularization phase, the separated spatially filtered signals are input into an independent component analysis mechanism that is configured with multi-tap filters, followed by secondary nonlinear spatial filters. Separated audio signals are the provided via an inverse-transform.

Abstract: A system and method for sound source separation. The system and method use a beamforming technique. The sound source separation system includes a windowing processor; a DFT transformer; a transfer function estimator; and a noise estimator. The system also includes a voice signal extractor that cancels individual voice signals, except an individual voice signal that is desired to be extracted among individual voice signals, from the integrated voice signals. The system further includes a voice signal detector that cancels a noise part provided through the noise estimator from a transfer function of an individual voice signal which is desired to be detected and extracts a noise-canceled individual voice signal. Even when two or more sound sources are simultaneously input, the sound sources can be separated from each other and separately stored and managed, or an initial sound source can be stored and managed.

Abstract: In one aspect thereof the invention provides a method for noise suppression of a speech signal that includes, for a speech signal having a frequency domain representation dividable into a plurality of frequency bins, determining a value of a scaling gain for at least some of said frequency bins and calculating smoothed scaling gain values. Calculating smoothed scaling gain values includes, for the at least some of the frequency bins, combining a currently determined value of the scaling gain and a previously determined value of the smoothed scaling gain. In another aspect a method partitions the plurality of frequency bins into a first set of contiguous frequency bins and a second set of contiguous frequency bins having a boundary frequency there between, where the boundary frequency differentiates between noise suppression techniques, and changes a value of the boundary frequency as a function of the spectral content of the speech signal.

Abstract: A method and apparatus of providing an audio output to a user in a communications system in which the audio to be output to a user, preferably an audio frame, is assessed before it is broadcast to the user, and then selectively changed on the basis of the assessment. The assessment may be carried out in the audio encoding process, in the audio decoding process and/or after the audio decoding process. The selective changing of the audio output may comprise selectively replacing the audio output and/or re-encoding of the audio output.

Abstract: A system and method for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient sound in the listening environment, comprises producing an audio sound in the time domain from an electrical sound signal in the time domain. The electrical sound signal in the time domain is transformed into an electrical sound signal in the frequency domain and the electrical sound signal in the frequency domain is retransformed into an audio sound in the time domain. The total sound level in the environment is measured and a signal representative thereof is generated. The audio sound signal and the total sound signal are processed to extract a signal representing the ambient sound level within the environment, and equalization is performed in the frequency domain to adjust the output from the audio sound signal to compensate for the ambient noise level.

Abstract: The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S?MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S?MIC).

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

Abstract: An audio signal may have a BL and an EL, wherein the EL represents additional information for enhancing the quality of the BL audio content. Decoding of such dual-layer signals usually comprises partial decoding of the BL data, wherein frequency bins of the BL are restored, mapping the restored frequency bins to the MDCT domain, adding them to the decoded EL and performing inverse Integer MDCT. A low-complexity method for decoding comprises reverse mapping of the decoded EL data, adding the reverse mapped EL data to the partially decoded BL data and filtering the sum, using the inverse BL filter bank.

Abstract: A voice activity detection process is robust to a low and high signal-to-noise ratio speech and signal loss. A process divides an aural signal into one or more bands. Signal magnitudes of frequency components and the respective noise components are estimated. A noise adaptation rate modifies estimates of noise components based on differences between the signal to the estimated noise and signal variability.

Abstract: An audio decoding device can adjust the high-range emphasis degree in accordance with a background noise level. The audio decoding device includes: a sound source signal decoder which performs a decoding process by using sound source encoding data separated by a separator so as to obtain a sound source signal; an LPC synthesis filter which performs an LPC synthesis filtering process by using a sound source signal and an LPC generated by an LPC decoder so as to obtain a decoded sound signal; a mode judger which determines whether a decoded sound signal is a stationary noise period by using a decoded LSP inputted from the LPC decoder a power calculator which calculates the power of the decoded audio signal; an SNR calculator which calculates an SNR of the decoded audio signal by using the power of the decoded audio signal and a mode judgment result in the mode judger and a post filter which performs a post filtering process by using the SNR of the decoded audio signal.

Abstract: Techniques described herein include the use of equalization techniques to improve intelligibility of a reproduced audio signal (e.g., a far-end speech signal).

Abstract: A noise reduction device of the present invention comprises a control filter unit for generating a control sound signal to cancel out a noise, a control speaker for outputting a control sound according to the control sound signal from the control filter unit, an error microphone for detecting a residual sound by superimposing the noise upon the control sound output from the control speaker, and an obstacle detector for detecting an obstacle around the error microphone, wherein the control filter unit generates the control sound signal according to data from the error microphone and the obstacle detector.

Abstract: Methods, systems, and apparatuses for performing packet loss concealment are disclosed. In response to determining that an encoded frame representing a segment of a signal is bad, an encoded parameter within the encoded frame is decoded based on bit information (such as soft bit information) associated with the encoded parameter to obtain a decoded parameter. Whether the decoded parameter violates a parameter constraint is determined. If a parameter constraint violation is detected, an estimate of the decoded parameter is generated. Either the decoded parameter or estimate of the decoded parameter is passed to a decoder for use in decoding the encoded frame.

Abstract: A noise reduction device is disclosed, in which noise reduction device, a controlling sound generator outputs a white noise generated by a white-noise generator, and this white noise is sensed by an error sensor for identifying an acoustic transmission function covering a path from the controlling sound generator to the error sensor. At this time, an identification controller prompts the white noise generator to generate a white noise for identifying the acoustic transmission function provided that an ambient noise level sensed by the error sensor is not greater than a given threshold.

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: A noise feedback coding (NFC) system and method that utilizes a simple and relatively inexpensive general structural configuration, but achieves improved flexibility with respect to controlling the shape of coding noise. The NFC system and method utilizes an all-zero noise feedback filter that is configured to approximate the response of a pole-zero noise feedback filter.

Abstract: An adaptive noise modeling speech recognition system improves speech recognition by modifying an activation of the system's grammar rules or models based on detected noise characteristics. An adaptive noise modeling speech recognition system includes a sensor that receives acoustic data having a speech component and a noise component. A processor analyzes the acoustic data and generates a noise indicator that identifies a characteristic of the noise component. An integrating decision logic processes the noise indicator and generates a noise model activation data structure that includes data that may be used by a speech recognition engine to adjust the activation of associated grammar rules or models.

Abstract: An apparatus for distinguishing a voice is described. In one embodiment, the apparatus includes a server with a communication interface, a frame generator, and a sound analyzer. The communication interface processes an incoming communication stream with an echo canceller to cancel echo in the communication stream. The frame generator operates on a processor and generates a plurality of frames from the communication stream. Each of the plurality of frames contains data for a period of time from the communication stream. The frame generator also assigns a frame value to each of the plurality of frames. The sound analyzer determines a status of the communication stream by analyzing the frame values of the plurality of frames.

Abstract: MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis.

Abstract: According to one embodiment, there is provided a sound quality control apparatus, including: a characteristic parameter extractor; a speech score calculator; a music score calculator; a power value acquisition module; a first storage configured to store speech scores and music scores; a second storage configured to store power values; a power-based score corrector configured to correct a current music score or a current speech score based on a first comparison result between a current power value and past power values, a second comparison result between the current music score and past music scores and a third comparison result between the current speech score and past speech scores; and a sound quality controller configured to perform a sound quality control by using at least one of the speech score and the music score corrected by the power-based score corrector.

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: Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders.

Abstract: Provided are a signal separating apparatus and a signal separating method capable of solving the permutation problem and separating user speech to be extracted. The signal separating apparatus separates a specific speech signal and a noise signal from a received sound signal. First, a joint probability density distribution estimation unit of a permutation solving unit calculates joint probability density distributions of the respective separated signals. Then, a classifying determination unit of the permutation solving unit determines classifying based on shapes of the calculated joint probability density distributions.

Abstract: The present technology provides a robust noise suppression system that may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. An acoustic signal may be received and transformed to cochlear domain sub-band signals. Features, such as pitch, may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals. An acoustic signal may be reconstructed from the noise-reduced sub-band signals.

Abstract: A decoding method and device are provided. The spectrum parameter of a current bad data frame is determined. Specifically, a number of continuous bad frames that occur currently is determined. A spectrum parameter of a good data frame before the current bad data frame is determined. And a constant mean value of a spectrum parameter is determined. Then, the spectrum parameter of the good data frame is adaptively shifted towards the constant mean value of the spectrum parameter according to the number of the continuous bad data frames to calculate and obtain spectrum parameter information of the current bad frame. When the continuous bad data frames occur, the relevance between the spectrum parameter of the nearest good frame and the spectrum parameter of the current bad frame is gradually reduced, so that more accurate spectrum parameter of the current bad data frame can be obtained, thereby obtaining a better speech quality under a same code rate and a same frame error rate.

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: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on a transition-band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: In one aspect, the invention provides an audio encoding method characterized by a decision being made as to whether the device which will decode the resulting bit stream Bitstream should apply post filtering including attenuation of interharmonic noise. Hence, the decision whether to use the post filter, which is encoded in the bit stream, is taken separately from the decision as to the most suitable coding mode. In another aspect, there is provided an audio decoding method with a decoding step followed by a post-filtering step, including interharmonic noise attenuation, and being characterized in a step of disabling the post filter in accordance with post filtering information encoded in the bit stream signal. Such a method is well suited for mixed-origin audio signals by virtue of its capability to deactivate the post filter in dependence of the post filtering information only, hence independently of factors such as the current coding mode.