Abstract: A method, an apparatus, and a computer program, which can suppress a low frequency range component with a small amount of calculation, and can achieve a noise suppression of high quality, are provided. The noise superposed in a desired signal of an input signal is suppressed by converting the input signal to a frequency domain signal; correcting an amplitude of the frequency domain signal to obtain an amplitude corrected signal; obtaining an estimated noise by using the amplitude corrected signal; determining a suppression coefficient by using the estimated noise and the amplitude corrected signal; and weighting the amplitude corrected signal with the suppression coefficient.

Abstract: The headset comprises: a physiological sensor suitable for being coupled to the cheek or the temple of the wearer of the headset and for picking up non-acoustic voice vibration transmitted by internal bone conduction; lowpass filter means for filtering the signal as picked up; a set of microphones picking up acoustic voice vibration transmitted by air from the mouth of the wearer of the headset; highpass filter means and noise-reduction means for acting on the signals picked up by the microphones; and mixer means for combining the filtered signals to output a signal representative of the speech uttered by the wearer of the headset. The signal of the physiological sensor is also used by means for calculating the cutoff frequency of the lowpass and highpass filters and by means for calculating the probability that speech is absent.

Abstract: A voice transmission apparatus is provided. The voice transmission apparatus includes a voice input unit, an Analog-to-Digital (AD) converter, a transmission synchronization unit, a channel status determination unit, and a transmission unit. The voice input unit receives an analog voice signal corresponding to a communication session. The AD converter converts the analog voice signal into a digital signal. The transmission synchronization unit generates a transmission target signal by respectively assigning a start identifier and an end identifier to the first frame and final frame of the digital signal corresponding to the communication session. The channel status determination unit determines the status of a channel. The transmission unit transmits the transmission target signal over the channel in such a way that the transmission rate of the transmission target signal is set on a communication session basis depending on the status of the channel.

Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.

Abstract: A system for combining signals includes a first microphone generating a first input signal having a first voice component and a first noise component, a second microphone generating a second input signal having a second voice component and a second noise component, a mixing circuit, and an adaptive filter. The mixing circuit applies a first gain having a value ? to the first input signal to produce a first scaled signal, applies a second gain having a value 1?? to the second input signal to produce a second scaled signal, and sums the first scaled signal and the second scaled signal to produce a summed signal. The adaptive filter computes an updated value of ? to minimize the energy of the summed signal based on the summed signal, the first input signal and the second input signal, and provides the updated value of ? to the mixing circuit.

Abstract: Electronic devices and accessories are provided that may communicate over wired communications paths. The electronic devices may be portable electronic devices such as cellular telephones or media players and may have audio connectors such as 3.5 mm audio jacks. The accessories may be headsets or other equipment having mating 3.5 mm audio plugs and speakers for playing audio. Microphones may be included in an accessory to gather voice signals and noise cancellation signals. Analog-to-digital converter circuitry in the accessory may digitize the microphone signals. Digital voice signals and voice noise cancellation signals can be transmitted over the communications path and processed by audio digital signal processor circuitry in an electronic device. Digital-to-analog converter circuitry in the accessory may convert digital audio signals to analog speaker signals.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: A device and a method for filtering out noise from speech of caller are disclosed. The method is applied to the device, includes: inputting a speech sound of a caller; converting the speech sound to digital signals by an analyzing-to-digital converting unit; analyzing the digital signals to identify a pure speech of the caller and filtering out an extraneous noise thus obtaining pure speech signals of the caller; encoding the pure speech signals by a coder and decoder unit, and submitting the encoded speech signals to the receiver.

Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

Abstract: A method, a device and a system for voice encoding/decoding are disclosed in the present invention. The method includes: assembling an input pulse code modulation signal into one signal according to a designated time slot and assembly manner; and encoding the assembled signal according to a designated encoding manner to output an encoded voice signal. In the present invention, because a process of assembling or splitting the signal may be implemented through software, in the case that hardware in a current network does not need to be replaced, an effect of encoding/decoding voice with a 7 K spectrum may be achieved in the current network.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the 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: A method for processing and iteratively enhancing and estimating a source audio signal received at two audio receivers is provided. In one embodiment, the method involves the use of codebook constrained iterative binaural Wiener filter (CCIBWF). The provided CCIBWF embodiment can improve the quality of speech received at two audio receivers both in terms of noise reduction and speech intelligibility. In one embodiment, optimum speech enhancement performance was achieved within two iterations of the CCIBWF scheme. Further, the embodiment of the CCIBWF scheme introduces minimal distortion to the binaural cues, such as the interaural time delay cues, thereby preserving localization information of the audio source. The embodiment of the CCIBWF is also able to relatively accurately track the Time Delay of Arrival (TDOA) when the audio source is moving. This ensures that the performance of the CCIBWF scheme is not significantly degraded due to the selection of wrong codebooks.

Abstract: A method of speech enhancement in a room (10), having the steps of: determining acoustic parameters of the room and a loudspeaker arrangement (24) located in the room, capturing audio signals from a speaker's voice with a microphone (12), and processing the captured audio signals with an audio signal processing unit (20). The audio signals are filtered by applying a selected frequency response curve to the audio signals, generating sound according to the processed audio signals by the loudspeaker arrangement, determining a value indicative of the overall gain applied to the captured audio signals, and selecting a frequency response curve to be applied to the captured audio signals according to the overall gain value and the acoustic parameters.

Abstract: The invention provides a method and device for enhancing the listening qualities of an audio file by providing the listener with a plurality of modified equalized audio files. Each modified equalized audio file having a consistent loudness level but different audio characteristics. Hence, for an input audio file the current invention allows the listener to individually select the best audio characteristics for them to listen to the content of the input audio file according to their particular requirements without them needing to adjust the loudness level in playback. The invention further enables the listener to switch between the multiple equalized audio files during playback. The invention further includes a SN detector and reducer to eliminate the adverse effects of the presence of sudden, strong noise in the input audio file in the process of generating the plurality of modified equalized audio files.

Abstract: Various embodiments relate to signal processing and, more particularly, to processing of received speech signals to preserve and enhance speech intelligibility. In one embodiment, a communications apparatus includes a receiving path over which received speech signals traverse in an audio stream, and an dynamic audio enhancement device disposed in the receiving path. The dynamic audio enhancement (“DAE”) device is configured to modify an amount of volume and an amount of equalization of the audio stream. The DAE device can include a noise level estimator (“NLE”) configured to generate a signal representing a noise level estimate. The noise level estimator can include a non-stationary noise detector and a stationary noise detector. The noise level estimator can be configured to generate the signal representing a first noise level estimate based on detection of the non-stationary noise or a second noise level estimate based on detection of the stationary noise.

Abstract: A system and method may receive a single-channel speech input captured via a microphone. For each current frame of speech input, the system and method may (a) perform a time-frequency transformation on the input signal over L (L>1) frames including the current frame to obtain an extended observation vector of the current frame, data elements in the extended observation vector representing the coefficients of the time-frequency transformation of the L frames of the speech input, (b) compute second-order statistics of the extended observation vector and of noise, and (c) construct a noise reduction filter for the current frame of the speech input based on the second-order statistics of the extended observation vector and the second-order statistics of noise.

Abstract: A system for and method of speech processing for a vehicle. Speech is received from at least one vehicle occupant via a plurality of microphones corresponding to the plurality of zones in the vehicle, wherein the microphones convert the speech into speech signals. At least one active communication zone is determined in which the at least one vehicle occupant corresponding to the active communication zone is speaking Speech processing is modified in response to the determined active communication zone.

Abstract: The present invention provides a noise reduction control method using a microphone array and a noise reduction control device using a microphone array wherein the method comprises the steps of: S1: collecting, by the microphone array, acoustic signals; S2: estimating incidence angles of all acoustic signals of the microphone array; S3: conducting a statistics on signal components according to incidence angles; S4: determining a parameter ? from a ratio of noise components according to the statistical result and using the parameter ? as a control parameter for controlling an adaptive filter. With the present invention, space position information of the sound is obtained directly with the microphone array to control update of the adaptive filter more accurately, so as to eliminate noise, enhance SNR and protect speech quality well at the same time.

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, an apparatus, and a computer program, which can suppress a low frequency range component with a small amount of calculation, and can achieve a noise suppression of high quality, are provided. The noise superposed in a desired signal of an input signal is suppressed by converting the input signal to a frequency domain signal; correcting an amplitude of the frequency domain signal to obtain an amplitude corrected signal; obtaining an estimated noise by using the amplitude corrected signal; determining a suppression coefficient by using the estimated noise and the amplitude corrected signal; and weighting the amplitude corrected signal with the suppression coefficient.

Abstract: Wind and other noise is suppressed in a signal by adaptively changing characteristics of a filter. The filter characteristics are changed in response to the noise content of the signal over time using a history of noise content. Filter characteristics are changed according to a plurality of reference filters, the characteristics of which are chosen to optimally attenuate or amplify signals in a range of frequencies.

Abstract: Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal.

Abstract: An apparatus is provided with a device storing machine readable code and a processor executing the machine readable code. The machine readable code includes sound setting code and audio processing code. The sound setting code detects use of a microphone and sets sound characteristics that are suitable for conversation in response to detecting the use of the microphone. The audio processing code processes sound on the basis of the sound characteristics set by the sound setting code.

Abstract: Systems and methods for adaptive intelligent noise suppression are provided. In exemplary embodiments, a primary acoustic signal is received. A speech distortion estimate is then determined based on the primary acoustic signal. The speech distortion estimate is used to derive control signals which adjust an enhancement filter. The enhancement filter is used to generate a plurality of gain masks, which may be applied to the primary acoustic signal to generate a noise suppressed signal.

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: Embodiments of the invention may provide the ability to selective filter sound from a voice communication, such as a telephone call, based on one or more attributes of the voice communication. Embodiments of the invention may select a filtering profile corresponding to the one or more attributes, and filter sound from the voice communication according to the selected profile. In one embodiment of the invention, the one or more attributes of the voice communication are determined from an electronic record corresponding to the voice communication, such as a calendar entry.

Abstract: Spectral components attenuated in a test denoised speech signal as a result of denoising a test speech signal are restored by representing a training undistorted speech signal as a composition of training undistorted bases, and representing a training denoised speech signal as a composition of training distorted bases. The test denoised signal decomposed as a composition of the training distorted bases. The undistorted test speech signal is then estimated as the composition of the training undistorted bases that is identical to the composition of training distorted bases.

Abstract: A method for reducing ringing in a signal output from a filter comprising inputting a signal into a filter; filtering a first portion of the input signal to generate a filtered portion of the output signal; analyzing the filtered portion of the output signal; detecting if ringing is present in the filtered portion of the output signal based on said analysis; and adjusting the filter characteristics to reduce ringing in a subsequent filtered portion of the output signal if it is determined that ringing is present.

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: Disclosed is a method of improving a sound quality, including: receiving a transmission signal of a first user equipment; removing noise in the transmission signal using noise information of the first user equipment side; performing speech reinforcement with respect to the noise removed transmission signal using noise information of a second user equipment side; and transmitting the speech reinforced transmission signal to the second user equipment.

Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

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: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: In order to reduce interference in an audio signal during a call on a mobile communication device, a plurality of transforms of the audio signal is performed, each transform containing phase information and amplitude information of corresponding samples of the audio signal. The results of the transforms are then averaged in order to generating a compensation signal that can be subtracted from the audio signal.

Abstract: Techniques are described herein that suppress noise in a Mel-filtered spectral domain. For example, a window may be applied to a representation of a speech signal in a time domain. The windowed representation in the time domain may be converted to a subsequent representation of the speech signal in the Mel-filtered spectral domain. A noise suppression operation may be performed with respect to the subsequent representation to provide noise-suppressed Mel coefficients.

Abstract: The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?1) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD'?1 Bw+(?2, PL,—using the signal quality measure in a quality estimation of the speech signal.

Abstract: Provided are methods and systems for enhancing the quality of voice communications. The method and corresponding system may involve classifying an audio signal into speech, and speech and noise and creating speech-noise classification data. The method may further involve sharing the speech-noise classification data with a speech encoder via a shared memory or by a Least Significant Bit (LSB) of a Pulse Code Modulation (PCM) stream. The method and corresponding system may also involve sharing acoustic cues with the speech encoder to improve the speech noise classification and, in certain embodiments, sharing scaling transition factors with the speech encoder to enable the speech encoder gradually change data rate in the transitions between the encoding modes.

Abstract: Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.

Abstract: A level of ambient noise at a local device is determined. A dynamic range compression (DRC) gain is computed based on the level of ambient noise at the local device. An additional gain factor is computed. A total gain is computed based on an adding of the DRC gain and the additional gain factor. An amplitude of an audio signal is adjusted based on the total gain, wherein the audio signal was transmitted from a remote device and received by the local device.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.

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: A method and an apparatus for processing an audio signal in a mobile terminal are provided, wherein an audio signal received from a counterpart mobile terminal is classified into a voice signal and a noise signal according to respective energy, and a frequency of the classified voice signal and an energy of the classified noise signal is controlled according to a predetermined criteria, then the controlled voice signal and the controlled noise signal are coupled and output to a speaker.

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: A howling canceller which suppresses occurrence of howling even when an open loop gain exceeds “1” in the whole reproduction band. In the howling canceller, an adaptive filter (107) operates a digital received voice signal with a tap coefficient to generate a pseudo echo; a subtractor (108) subtracts the pseudo echo from a digital transmitted voice signal to generate a residual signal; and an amplitude limiting circuit (110) limits the absolute value of the amplitude of the digital received voice signal to be equal to or smaller than a predetermined threshold which ensures that all of a D/A converter (101), a power amplifier (102), a speaker (103), a microphone (104), a microphone amplifier (105), and an A/D converter (106) operate in a linear operation area, and outputs the amplitude-limited digital received voice signal to the D/A converter (101) and the adaptive filter (107).

Abstract: Acoustic noise suppression is provided in multiple-microphone systems using Voice Activity Detectors (VAD). A host system receives acoustic signals via multiple microphones. The system also receives information on the vibration of human tissue associated with human voicing activity via the VAD. In response, the system generates a transfer function representative of the received acoustic signals upon determining that voicing information is absent from the received acoustic signals during at least one specified period of time. The system removes noise from the received acoustic signals using the transfer function, thereby producing a denoised acoustic data stream.

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.

Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.

Abstract: A speech enhancement system improves the speech quality and intelligibility of a speech signal. The system includes a time-to-frequency converter that converts segments of a speech signal into frequency bands. A signal detector measures the signal power of the frequency bands of each speech segment. A background noise estimator measures a background noise detected in the speech signal. A dynamic noise reduction controller dynamically models the background noise in the speech signal. The speech enhancement renders a speech signal perceptually pleasing to a listener by dynamically attenuating a portion of the noise that occurs in a portion of the spectrum of the speech signal.